Bug 1766646 (MOZ) handle upstream adding {Audio|Video}ReceiveStream::transport_cc and removing {Audio|Video}ReceiveStream::rtp_config

This commit is contained in:
Michael Froman 2022-07-14 21:45:44 -05:00 committed by Connor Sheehan
parent 2dfb4376ae
commit a79102a418
2 changed files with 2 additions and 12 deletions

View File

@ -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<int, webrtc::SdpAudioFormat> 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<webrtc::RtpExtension>& MockVideoReceiveStream::GetRtpExtensions() const {
static std::vector<webrtc::RtpExtension> rtpExtensions;
return rtpExtensions;

View File

@ -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; }