Reland: Remove unsupported configuration value, `allow_codec_switching`
This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995.
Reason for revert: Relanding once downstream issues have been addressed
Original change's description:
> Revert "Remove unsupported configuration value, `allow_codec_switching`"
>
> This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023.
>
> Reason for revert: breaks downstream
>
> Original change's description:
> > Remove unsupported configuration value, `allow_codec_switching`
> >
> > Bug: webrtc:11341
> > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
> > Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40995}
>
> Bug: webrtc:11341
> Change-Id: I784fd95062fc71f8dcc139b05121985f60709004
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780
> Owners-Override: Philip Eliasson <philipel@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40998}
Bug: webrtc:11341
Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41032}
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 275daa6..9243190 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -661,10 +661,6 @@
// Added to be able to control rollout of this feature.
bool enable_implicit_rollback = false;
- // Whether network condition based codec switching is allowed.
- // TODO(bugs.webrtc.org/11341): Remove this unsupported config value.
- absl::optional<bool> allow_codec_switching;
-
// The delay before doing a usage histogram report for long-lived
// PeerConnections. Used for testing only.
absl::optional<int> report_usage_pattern_delay_ms;
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 046e08f..183cbeb 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -329,7 +329,6 @@
modified_config.active_reset_srtp_params =
configuration.active_reset_srtp_params;
modified_config.turn_logging_id = configuration.turn_logging_id;
- modified_config.allow_codec_switching = configuration.allow_codec_switching;
modified_config.stable_writable_connection_ping_interval_ms =
configuration.stable_writable_connection_ping_interval_ms;
if (configuration != modified_config) {
@@ -458,7 +457,6 @@
bool offer_extmap_allow_mixed;
std::string turn_logging_id;
bool enable_implicit_rollback;
- absl::optional<bool> allow_codec_switching;
absl::optional<int> report_usage_pattern_delay_ms;
absl::optional<int> stable_writable_connection_ping_interval_ms;
webrtc::VpnPreference vpn_preference;
@@ -522,7 +520,6 @@
offer_extmap_allow_mixed == o.offer_extmap_allow_mixed &&
turn_logging_id == o.turn_logging_id &&
enable_implicit_rollback == o.enable_implicit_rollback &&
- allow_codec_switching == o.allow_codec_switching &&
report_usage_pattern_delay_ms == o.report_usage_pattern_delay_ms &&
stable_writable_connection_ping_interval_ms ==
o.stable_writable_connection_ping_interval_ms &&
diff --git a/sdk/android/api/org/webrtc/PeerConnection.java b/sdk/android/api/org/webrtc/PeerConnection.java
index 5c87fe3..d530bc2 100644
--- a/sdk/android/api/org/webrtc/PeerConnection.java
+++ b/sdk/android/api/org/webrtc/PeerConnection.java
@@ -540,11 +540,6 @@
// every offer/answer negotiation.This is only intended to be a workaround for crbug.com/835958
public boolean activeResetSrtpParams;
- // Whether this client is allowed to switch encoding codec mid-stream. This is a workaround for
- // a WebRTC bug where the receiver could get confussed if a codec switch happened mid-call.
- // Null indicates no change to currently configured value.
- @Nullable public Boolean allowCodecSwitching;
-
/**
* Defines advanced optional cryptographic settings related to SRTP and
* frame encryption for native WebRTC. Setting this will overwrite any
@@ -611,7 +606,6 @@
activeResetSrtpParams = false;
cryptoOptions = null;
turnLoggingId = null;
- allowCodecSwitching = null;
enableImplicitRollback = false;
offerExtmapAllowMixed = true;
}
@@ -803,12 +797,6 @@
@Nullable
@CalledByNative("RTCConfiguration")
- Boolean getAllowCodecSwitching() {
- return allowCodecSwitching;
- }
-
- @Nullable
- @CalledByNative("RTCConfiguration")
CryptoOptions getCryptoOptions() {
return cryptoOptions;
}
diff --git a/sdk/android/src/jni/pc/peer_connection.cc b/sdk/android/src/jni/pc/peer_connection.cc
index a063804..e5fb11e 100644
--- a/sdk/android/src/jni/pc/peer_connection.cc
+++ b/sdk/android/src/jni/pc/peer_connection.cc
@@ -267,10 +267,6 @@
Java_RTCConfiguration_getActiveResetSrtpParams(jni, j_rtc_config);
rtc_config->crypto_options =
JavaToNativeOptionalCryptoOptions(jni, j_crypto_options);
-
- rtc_config->allow_codec_switching = JavaToNativeOptionalBool(
- jni, Java_RTCConfiguration_getAllowCodecSwitching(jni, j_rtc_config));
-
rtc_config->offer_extmap_allow_mixed =
Java_RTCConfiguration_getOfferExtmapAllowMixed(jni, j_rtc_config);
rtc_config->enable_implicit_rollback =
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.h b/sdk/objc/api/peerconnection/RTCConfiguration.h
index 1b0d14b..011eaa6 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.h
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.h
@@ -184,12 +184,6 @@
*/
@property(nonatomic, assign) BOOL activeResetSrtpParams;
-/** If the remote side support mid-stream codec switches then allow encoder
- * switching to be performed.
- */
-
-@property(nonatomic, assign) BOOL allowCodecSwitching;
-
/**
* Defines advanced optional cryptographic settings related to SRTP and
* frame encryption for native WebRTC. Setting this will overwrite any
diff --git a/sdk/objc/api/peerconnection/RTCConfiguration.mm b/sdk/objc/api/peerconnection/RTCConfiguration.mm
index 8e42cb2..86ecbab 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.mm
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -51,7 +51,6 @@
@synthesize sdpSemantics = _sdpSemantics;
@synthesize turnCustomizer = _turnCustomizer;
@synthesize activeResetSrtpParams = _activeResetSrtpParams;
-@synthesize allowCodecSwitching = _allowCodecSwitching;
@synthesize cryptoOptions = _cryptoOptions;
@synthesize turnLoggingId = _turnLoggingId;
@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
@@ -139,7 +138,6 @@
_turnLoggingId = [NSString stringWithUTF8String:config.turn_logging_id.c_str()];
_rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
_rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
- _allowCodecSwitching = config.allow_codec_switching.value_or(false);
_enableImplicitRollback = config.enable_implicit_rollback;
_offerExtmapAllowMixed = config.offer_extmap_allow_mixed;
_iceCheckIntervalStrongConnectivity =
@@ -286,7 +284,6 @@
nativeConfig->turn_logging_id = [_turnLoggingId UTF8String];
nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
nativeConfig->set_video_rtcp_report_interval_ms(_rtcpVideoReportIntervalMs);
- nativeConfig->allow_codec_switching = _allowCodecSwitching;
nativeConfig->enable_implicit_rollback = _enableImplicitRollback;
nativeConfig->offer_extmap_allow_mixed = _offerExtmapAllowMixed;
if (_iceCheckIntervalStrongConnectivity != nil) {
