imrpove: remove unused definition

Author: yeoleobun

tests/sdp_bridge_e2e_test.rs

@@ -13,62 +13,6 @@ use rustpbx::media::{RtpTrackBuilder, Track};
 use rustrtc::TransportMode;
 use rustrtc::sdp::{SdpType, SessionDescription};
 
-struct BridgeCodecLists {
-    caller_side: Vec<rustpbx::media::negotiate::CodecInfo>,
-    callee_side: Vec<rustpbx::media::negotiate::CodecInfo>,
-}
-
-trait TestBridgeCodecLists {
-    fn build_bridge_codec_lists(
-        caller_sdp: &str,
-        caller_is_webrtc: bool,
-        callee_is_webrtc: bool,
-        allow_codecs: &[CodecType],
-    ) -> BridgeCodecLists;
-}
-
-impl TestBridgeCodecLists for rustpbx::media::negotiate::MediaNegotiator {
-    fn build_bridge_codec_lists(
-        caller_sdp: &str,
-        caller_is_webrtc: bool,
-        callee_is_webrtc: bool,
-        allow_codecs: &[CodecType],
-    ) -> BridgeCodecLists {
-        let mut caller_side = Self::build_codec_list_from_offer(caller_sdp, allow_codecs);
-        if caller_is_webrtc {
-            caller_side = filter_webrtc_answer_codecs(caller_side);
-        }
-
-        let mut callee_side = Self::build_callee_codec_offer_with_allow(caller_sdp, allow_codecs);
-        if callee_is_webrtc {
-            callee_side = Self::filter_webrtc_offer_codecs(caller_sdp, callee_side);
-        }
-
-        BridgeCodecLists {
-            caller_side,
-            callee_side,
-        }
-    }
-}
-
-fn filter_webrtc_answer_codecs(
-    codecs: Vec<rustpbx::media::negotiate::CodecInfo>,
-) -> Vec<rustpbx::media::negotiate::CodecInfo> {
-    let audio_clock_rates: std::collections::HashSet<_> = codecs
-        .iter()
-        .filter(|codec| !codec.is_dtmf() && codec.codec != CodecType::G729)
-        .map(|codec| codec.clock_rate)
-        .collect();
-
-    codecs
-        .into_iter()
-        .filter(|codec| {
-            codec.codec != CodecType::G729
-                && (!codec.is_dtmf() || audio_clock_rates.contains(&codec.clock_rate))
-        })
-        .collect()
-}
-
 /// Helper: create a WebRTC track simulating a browser caller (e.g., JsSIP).
 fn create_webrtc_caller(port_start: u16) -> Box<dyn Track> {
     Box::new(