Remove unused combined_audio_video_bwe.
Bug: None
Change-Id: Ie539351f98b7a0ebb5f08e0df5c5759a2bcb5588
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40160}
diff --git a/api/audio_options.cc b/api/audio_options.cc
index 6585150..a3f2b6e 100644
--- a/api/audio_options.cc
+++ b/api/audio_options.cc
@@ -52,7 +52,6 @@
change.audio_jitter_buffer_fast_accelerate);
SetFrom(&audio_jitter_buffer_min_delay_ms,
change.audio_jitter_buffer_min_delay_ms);
- SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
SetFrom(&init_recording_on_send, change.init_recording_on_send);
@@ -72,7 +71,6 @@
o.audio_jitter_buffer_fast_accelerate &&
audio_jitter_buffer_min_delay_ms ==
o.audio_jitter_buffer_min_delay_ms &&
- combined_audio_video_bwe == o.combined_audio_video_bwe &&
audio_network_adaptor == o.audio_network_adaptor &&
audio_network_adaptor_config == o.audio_network_adaptor_config &&
init_recording_on_send == o.init_recording_on_send;
@@ -97,7 +95,6 @@
audio_jitter_buffer_fast_accelerate);
ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms",
audio_jitter_buffer_min_delay_ms);
- ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);
result << "}";
diff --git a/api/audio_options.h b/api/audio_options.h
index 39ba388..3ab3b3c 100644
--- a/api/audio_options.h
+++ b/api/audio_options.h
@@ -58,11 +58,6 @@
absl::optional<bool> audio_jitter_buffer_fast_accelerate;
// Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds.
absl::optional<int> audio_jitter_buffer_min_delay_ms;
- // Enable combined audio+bandwidth BWE.
- // TODO(pthatcher): This flag is set from the
- // "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
- // and check if any other AudioOptions members are unused.
- absl::optional<bool> combined_audio_video_bwe;
// Enable audio network adaptor.
// TODO(webrtc:11717): Remove this API in favor of adaptivePtime in
// RtpEncodingParameters.
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 6ce4650..e80550c 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -448,9 +448,6 @@
// when switching from a static scene to one with motion.
absl::optional<int> screencast_min_bitrate;
- // Use new combined audio/video bandwidth estimation?
- absl::optional<bool> combined_audio_video_bwe;
-
#if defined(WEBRTC_FUCHSIA)
// TODO(bugs.webrtc.org/11066): Remove entirely once Fuchsia does not use.
// TODO(bugs.webrtc.org/9891) - Move to crypto_options
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 59d3872..e0cef06 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -303,7 +303,6 @@
int max_ipv6_networks;
bool disable_link_local_networks;
absl::optional<int> screencast_min_bitrate;
- absl::optional<bool> combined_audio_video_bwe;
#if defined(WEBRTC_FUCHSIA)
absl::optional<bool> enable_dtls_srtp;
#endif
@@ -372,7 +371,6 @@
max_ipv6_networks == o.max_ipv6_networks &&
disable_link_local_networks == o.disable_link_local_networks &&
screencast_min_bitrate == o.screencast_min_bitrate &&
- combined_audio_video_bwe == o.combined_audio_video_bwe &&
#if defined(WEBRTC_FUCHSIA)
enable_dtls_srtp == o.enable_dtls_srtp &&
#endif
diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc
index 56ff8ff..2213db6 100644
--- a/pc/peer_connection_media_unittest.cc
+++ b/pc/peer_connection_media_unittest.cc
@@ -1348,21 +1348,6 @@
"Failed to set remote offer sdp: Duplicate a=mid value 'same'.");
}
-TEST_P(PeerConnectionMediaTest,
- CombinedAudioVideoBweConfigPropagatedToMediaEngine) {
- RTCConfiguration config;
- config.combined_audio_video_bwe.emplace(true);
- auto caller = CreatePeerConnectionWithAudioVideo(config);
-
- ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
-
- auto caller_voice = caller->media_engine()->GetVoiceSendChannel(0);
- ASSERT_TRUE(caller_voice);
- const cricket::AudioOptions& audio_options = caller_voice->options();
- EXPECT_EQ(config.combined_audio_video_bwe,
- audio_options.combined_audio_video_bwe);
-}
-
// Test that if a RED codec refers to another codec in its fmtp line, but that
// codec's payload type was reassigned for some reason (either the remote
// endpoint selected a different payload type or there was a conflict), the RED
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 7f3db91..c87b6ec 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -1326,8 +1326,6 @@
// RTCConfiguration value (not available on Web).
video_options_.screencast_min_bitrate_kbps =
configuration.screencast_min_bitrate.value_or(100);
- audio_options_.combined_audio_video_bwe =
- configuration.combined_audio_video_bwe;
audio_options_.audio_jitter_buffer_max_packets =
configuration.audio_jitter_buffer_max_packets;
diff --git a/sdk/android/api/org/webrtc/PeerConnection.java b/sdk/android/api/org/webrtc/PeerConnection.java
index e334a3f..5c87fe3 100644
--- a/sdk/android/api/org/webrtc/PeerConnection.java
+++ b/sdk/android/api/org/webrtc/PeerConnection.java
@@ -528,7 +528,6 @@
public boolean enableCpuOveruseDetection;
public boolean suspendBelowMinBitrate;
@Nullable public Integer screencastMinBitrate;
- @Nullable public Boolean combinedAudioVideoBwe;
// Use "Unknown" to represent no preference of adapter types, not the
// preference of adapters of unknown types.
public AdapterType networkPreference;
@@ -607,7 +606,6 @@
enableCpuOveruseDetection = true;
suspendBelowMinBitrate = false;
screencastMinBitrate = null;
- combinedAudioVideoBwe = null;
networkPreference = AdapterType.UNKNOWN;
sdpSemantics = SdpSemantics.UNIFIED_PLAN;
activeResetSrtpParams = false;
@@ -788,12 +786,6 @@
return screencastMinBitrate;
}
- @Nullable
- @CalledByNative("RTCConfiguration")
- Boolean getCombinedAudioVideoBwe() {
- return combinedAudioVideoBwe;
- }
-
@CalledByNative("RTCConfiguration")
AdapterType getNetworkPreference() {
return networkPreference;
diff --git a/sdk/android/src/jni/pc/peer_connection.cc b/sdk/android/src/jni/pc/peer_connection.cc
index 9983ae7..a063804 100644
--- a/sdk/android/src/jni/pc/peer_connection.cc
+++ b/sdk/android/src/jni/pc/peer_connection.cc
@@ -260,8 +260,6 @@
Java_RTCConfiguration_getSuspendBelowMinBitrate(jni, j_rtc_config);
rtc_config->screencast_min_bitrate = JavaToNativeOptionalInt(
jni, Java_RTCConfiguration_getScreencastMinBitrate(jni, j_rtc_config));
- rtc_config->combined_audio_video_bwe = JavaToNativeOptionalBool(
- jni, Java_RTCConfiguration_getCombinedAudioVideoBwe(jni, j_rtc_config));
rtc_config->network_preference =
JavaToNativeNetworkPreference(jni, j_network_preference);
rtc_config->sdp_semantics = JavaToNativeSdpSemantics(jni, j_sdp_semantics);
diff --git a/sdk/media_constraints.cc b/sdk/media_constraints.cc
index bbb46ed..88261e7 100644
--- a/sdk/media_constraints.cc
+++ b/sdk/media_constraints.cc
@@ -117,8 +117,6 @@
const char MediaConstraints::kEnableDscp[] = "googDscp";
const char MediaConstraints::kEnableVideoSuspendBelowMinBitrate[] =
"googSuspendBelowMinBitrate";
-const char MediaConstraints::kCombinedAudioVideoBwe[] =
- "googCombinedAudioVideoBwe";
const char MediaConstraints::kScreencastMinBitrate[] =
"googScreencastMinBitrate";
// TODO(ronghuawu): Remove once cpu overuse detection is stable.
@@ -162,9 +160,6 @@
ConstraintToOptional<int>(constraints,
MediaConstraints::kScreencastMinBitrate,
&configuration->screencast_min_bitrate);
- ConstraintToOptional<bool>(constraints,
- MediaConstraints::kCombinedAudioVideoBwe,
- &configuration->combined_audio_video_bwe);
}
void CopyConstraintsIntoAudioOptions(const MediaConstraints* constraints,
diff --git a/sdk/media_constraints.h b/sdk/media_constraints.h
index a428abd..5bd38c2 100644
--- a/sdk/media_constraints.h
+++ b/sdk/media_constraints.h
@@ -89,9 +89,6 @@
static const char kEnableIPv6[]; // googIPv6
// Temporary constraint to enable suspend below min bitrate feature.
static const char kEnableVideoSuspendBelowMinBitrate[];
- // googSuspendBelowMinBitrate
- // Constraint to enable combined audio+video bandwidth estimation.
- static const char kCombinedAudioVideoBwe[]; // googCombinedAudioVideoBwe
static const char kScreencastMinBitrate[]; // googScreencastMinBitrate
static const char kCpuOveruseDetection[]; // googCpuOveruseDetection
diff --git a/sdk/media_constraints_unittest.cc b/sdk/media_constraints_unittest.cc
index 2d25da0..5e6b157 100644
--- a/sdk/media_constraints_unittest.cc
+++ b/sdk/media_constraints_unittest.cc
@@ -23,7 +23,6 @@
return a.audio_jitter_buffer_max_packets ==
b.audio_jitter_buffer_max_packets &&
a.screencast_min_bitrate == b.screencast_min_bitrate &&
- a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
a.media_config == b.media_config;
}
