Delete unused key WebRTC-Audio-SendSideBwe-For-Video.
Bug: webrtc:10286
Change-Id: If9ddbe71d9ba1afe51be5f9f46fcd4a72b34bc7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123787
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26990}diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 8157e6a..d3ec157 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -569,9 +569,7 @@
// 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 @@
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 a505357..a601cce 100644
--- a/rtc_base/experiments/audio_allocation_settings.cc
+++ b/rtc_base/experiments/audio_allocation_settings.cc
@@ -22,7 +22,6 @@
: 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 @@
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 @@
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 f05b4a3..32e11df 100644
--- a/rtc_base/experiments/audio_allocation_settings.h
+++ b/rtc_base/experiments/audio_allocation_settings.h
@@ -34,12 +34,6 @@
// 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 @@
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 cf648d3..61443e8 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 127ce6f..4a30db2 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";
