Revert "Reland: Remove unsupported configuration value, `allow_codec_switching`"
This reverts commit 23501a2aa656b94e26d4c67b8b9393258551560f.
Reason for revert: Breaks downstream features
Original change's description:
> 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}
Bug: webrtc:11341
Change-Id: I0eb8e6a464a8a51e6359caf8f43231dc275c4f20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327382
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41161}
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 8977137..2d0abd8 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -666,6 +666,10 @@
// 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 46c28bb..8afc44e 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -329,6 +329,7 @@
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) {
@@ -457,6 +458,7 @@
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;
VpnPreference vpn_preference;
@@ -520,6 +522,7 @@
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 d530bc2..5c87fe3 100644
--- a/sdk/android/api/org/webrtc/PeerConnection.java
+++ b/sdk/android/api/org/webrtc/PeerConnection.java
@@ -540,6 +540,11 @@
// 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
@@ -606,6 +611,7 @@
activeResetSrtpParams = false;
cryptoOptions = null;
turnLoggingId = null;
+ allowCodecSwitching = null;
enableImplicitRollback = false;
offerExtmapAllowMixed = true;
}
@@ -797,6 +803,12 @@
@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 e5fb11e..a063804 100644
--- a/sdk/android/src/jni/pc/peer_connection.cc
+++ b/sdk/android/src/jni/pc/peer_connection.cc
@@ -267,6 +267,10 @@
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 011eaa6..1b0d14b 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.h
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.h
@@ -184,6 +184,12 @@
*/
@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 86ecbab..8e42cb2 100644
--- a/sdk/objc/api/peerconnection/RTCConfiguration.mm
+++ b/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -51,6 +51,7 @@
@synthesize sdpSemantics = _sdpSemantics;
@synthesize turnCustomizer = _turnCustomizer;
@synthesize activeResetSrtpParams = _activeResetSrtpParams;
+@synthesize allowCodecSwitching = _allowCodecSwitching;
@synthesize cryptoOptions = _cryptoOptions;
@synthesize turnLoggingId = _turnLoggingId;
@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
@@ -138,6 +139,7 @@
_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 =
@@ -284,6 +286,7 @@
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) {
