From a79102a41815cf6bc35bea1501840e8fd86e27c6 Mon Sep 17 00:00:00 2001 From: Michael Froman Date: Thu, 14 Jul 2022 21:45:44 -0500 Subject: [PATCH] Bug 1766646 (MOZ) handle upstream adding {Audio|Video}ReceiveStream::transport_cc and removing {Audio|Video}ReceiveStream::rtp_config --- media/webrtc/signaling/gtest/MockCall.cpp | 10 ---------- media/webrtc/signaling/gtest/MockCall.h | 4 ++-- 2 files changed, 2 insertions(+), 12 deletions(-) diff --git a/media/webrtc/signaling/gtest/MockCall.cpp b/media/webrtc/signaling/gtest/MockCall.cpp index 958f7e5c0da9..95075523eb85 100644 --- a/media/webrtc/signaling/gtest/MockCall.cpp +++ b/media/webrtc/signaling/gtest/MockCall.cpp @@ -14,11 +14,6 @@ void MockAudioSendStream::Reconfigure(const Config& config) { mCallWrapper->GetMockCall()->mAudioSendConfig = mozilla::Some(config); } -const webrtc::ReceiveStream::RtpConfig& MockAudioReceiveStream::rtp_config() const { - MOZ_ASSERT(mCallWrapper->GetMockCall()->mAudioReceiveConfig.isSome()); - return mCallWrapper->GetMockCall()->mAudioReceiveConfig->rtp; -} - void MockAudioReceiveStream::SetDecoderMap( std::map decoder_map) { MOZ_ASSERT(mCallWrapper->GetMockCall()->mAudioReceiveConfig.isSome()); @@ -45,11 +40,6 @@ void MockVideoSendStream::ReconfigureVideoEncoder( mozilla::Some(config.Copy()); } -const webrtc::ReceiveStream::RtpConfig& MockVideoReceiveStream::rtp_config() const { - MOZ_ASSERT(mCallWrapper->GetMockCall()->mVideoReceiveConfig.isSome()); - return mCallWrapper->GetMockCall()->mVideoReceiveConfig->rtp; -} - const std::vector& MockVideoReceiveStream::GetRtpExtensions() const { static std::vector rtpExtensions; return rtpExtensions; diff --git a/media/webrtc/signaling/gtest/MockCall.h b/media/webrtc/signaling/gtest/MockCall.h index 875b725ca1b2..8d953c5667ac 100644 --- a/media/webrtc/signaling/gtest/MockCall.h +++ b/media/webrtc/signaling/gtest/MockCall.h @@ -63,7 +63,7 @@ class MockAudioReceiveStream : public webrtc::AudioReceiveStream { bool IsRunning() const override { return true; } - const webrtc::ReceiveStream::RtpConfig& rtp_config() const override; + bool transport_cc() const override { return false; } Stats GetStats(bool get_and_clear_legacy_stats) const override { return mStats; @@ -153,7 +153,7 @@ class MockVideoReceiveStream : public webrtc::VideoReceiveStream { void Stop() override {} - const webrtc::ReceiveStream::RtpConfig& rtp_config() const override; + bool transport_cc() const override { return false; } Stats GetStats() const override { return mStats; }