From 110c64bcd63943d69142fd8580d65d08fbb93381 Mon Sep 17 00:00:00 2001 From: Christoffer Rodbro Date: Wed, 6 Mar 2019 09:51:08 +0100 Subject: [PATCH] Delete unused key WebRTC-Audio-SendSideBwe-For-Video. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:10286 Change-Id: If9ddbe71d9ba1afe51be5f9f46fcd4a72b34bc7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123787 Reviewed-by: Sebastian Jansson Reviewed-by: Kári Helgason Reviewed-by: Oskar Sundbom Commit-Queue: Christoffer Rodbro Cr-Commit-Position: refs/heads/master@{#26990} --- audio/audio_send_stream.cc | 8 ++------ .../experiments/audio_allocation_settings.cc | 17 ----------------- .../experiments/audio_allocation_settings.h | 7 ------- sdk/objc/api/peerconnection/RTCFieldTrials.h | 1 - sdk/objc/api/peerconnection/RTCFieldTrials.mm | 1 - 5 files changed, 2 insertions(+), 32 deletions(-) diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index 8157e6a324..d3ec157849 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -569,9 +569,7 @@ bool AudioSendStream::SetupSendCodec(AudioSendStream* stream, // If a bitrate has been specified for the codec, use it over the // codec's default. - if (stream->allocation_settings_.UpdateAudioTargetBitrate( - TransportSeqNumId(new_config)) && - spec.target_bitrate_bps) { + if (spec.target_bitrate_bps) { encoder->OnReceivedTargetAudioBitrate(*spec.target_bitrate_bps); } @@ -646,9 +644,7 @@ bool AudioSendStream::ReconfigureSendCodec(AudioSendStream* stream, new_config.send_codec_spec->target_bitrate_bps; // If a bitrate has been specified for the codec, use it over the // codec's default. - if (stream->allocation_settings_.UpdateAudioTargetBitrate( - TransportSeqNumId(new_config)) && - new_target_bitrate_bps && + if (new_target_bitrate_bps && new_target_bitrate_bps != old_config.send_codec_spec->target_bitrate_bps) { stream->channel_send_->CallEncoder([&](AudioEncoder* encoder) { diff --git a/rtc_base/experiments/audio_allocation_settings.cc b/rtc_base/experiments/audio_allocation_settings.cc index a505357869..a601cce340 100644 --- a/rtc_base/experiments/audio_allocation_settings.cc +++ b/rtc_base/experiments/audio_allocation_settings.cc @@ -22,7 +22,6 @@ AudioAllocationSettings::AudioAllocationSettings() : audio_send_side_bwe_("Enabled"), allocate_audio_without_feedback_("Enabled"), force_no_audio_feedback_("Enabled"), - audio_feedback_to_improve_video_bwe_("Enabled"), send_side_bwe_with_overhead_("Enabled"), default_min_bitrate_("min", DataRate::bps(kOpusMinBitrateBps)), default_max_bitrate_("max", DataRate::bps(kOpusBitrateFbBps)), @@ -33,9 +32,6 @@ AudioAllocationSettings::AudioAllocationSettings() field_trial::FindFullName("WebRTC-Audio-ABWENoTWCC")); ParseFieldTrial({&force_no_audio_feedback_}, field_trial::FindFullName("WebRTC-Audio-ForceNoTWCC")); - ParseFieldTrial( - {&audio_feedback_to_improve_video_bwe_}, - field_trial::FindFullName("WebRTC-Audio-SendSideBwe-For-Video")); ParseFieldTrial({&send_side_bwe_with_overhead_}, field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead")); @@ -73,19 +69,6 @@ bool AudioAllocationSettings::ShouldSendTransportSequenceNumber( transport_seq_num_extension_header_id != 0; } -bool AudioAllocationSettings::UpdateAudioTargetBitrate( - int transport_seq_num_extension_header_id) const { - // If other side does not support audio TWCC and WebRTC-Audio-ABWENoTWCC is - // not enabled, do not update target audio bitrate if we are in - // WebRTC-Audio-SendSideBwe-For-Video experiment - if (allocate_audio_without_feedback_ || - transport_seq_num_extension_header_id != 0) - return true; - if (audio_feedback_to_improve_video_bwe_) - return false; - return true; -} - bool AudioAllocationSettings::IncludeAudioInAllocationOnStart( int min_bitrate_bps, int max_bitrate_bps, diff --git a/rtc_base/experiments/audio_allocation_settings.h b/rtc_base/experiments/audio_allocation_settings.h index f05b4a33ed..32e11df46e 100644 --- a/rtc_base/experiments/audio_allocation_settings.h +++ b/rtc_base/experiments/audio_allocation_settings.h @@ -34,12 +34,6 @@ class AudioAllocationSettings { // configured. bool ShouldSendTransportSequenceNumber( int transport_seq_num_extension_header_id) const; - // Returns true if target bitrate for audio streams should be updated. - // |transport_seq_num_extension_header_id| the extension header id for - // transport sequence numbers. Set to 0 if not the extension is not - // configured. - bool UpdateAudioTargetBitrate( - int transport_seq_num_extension_header_id) const; // Returns true if audio should be added to rate allocation when the audio // stream is started. // |min_bitrate_bps| the configured min bitrate, set to -1 if unset. @@ -83,7 +77,6 @@ class AudioAllocationSettings { FieldTrialFlag audio_send_side_bwe_; FieldTrialFlag allocate_audio_without_feedback_; FieldTrialFlag force_no_audio_feedback_; - FieldTrialFlag audio_feedback_to_improve_video_bwe_; FieldTrialFlag send_side_bwe_with_overhead_; int min_overhead_bps_ = 0; // Default bitrates to use as range if there's no user configured diff --git a/sdk/objc/api/peerconnection/RTCFieldTrials.h b/sdk/objc/api/peerconnection/RTCFieldTrials.h index cf648d32c3..61443e8bb2 100644 --- a/sdk/objc/api/peerconnection/RTCFieldTrials.h +++ b/sdk/objc/api/peerconnection/RTCFieldTrials.h @@ -14,7 +14,6 @@ /** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */ RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweKey; -RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweForVideoKey; RTC_EXTERN NSString * const kRTCFieldTrialAudioForceNoTWCCKey; RTC_EXTERN NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey; RTC_EXTERN NSString * const kRTCFieldTrialSendSideBweWithOverheadKey; diff --git a/sdk/objc/api/peerconnection/RTCFieldTrials.mm b/sdk/objc/api/peerconnection/RTCFieldTrials.mm index 127ce6feb8..4a30db2f70 100644 --- a/sdk/objc/api/peerconnection/RTCFieldTrials.mm +++ b/sdk/objc/api/peerconnection/RTCFieldTrials.mm @@ -17,7 +17,6 @@ #include "system_wrappers/include/field_trial.h" NSString * const kRTCFieldTrialAudioSendSideBweKey = @"WebRTC-Audio-SendSideBwe"; -NSString * const kRTCFieldTrialAudioSendSideBweForVideoKey = @"WebRTC-Audio-SendSideBwe-For-Video"; NSString * const kRTCFieldTrialAudioForceNoTWCCKey = @"WebRTC-Audio-ForceNoTWCC"; NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey = @"WebRTC-Audio-ABWENoTWCC"; NSString * const kRTCFieldTrialSendSideBweWithOverheadKey = @"WebRTC-SendSideBwe-WithOverhead";