Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index bdff1e70f3a8b40100c77f4f5bc1a92c34dcaf0f..34ecfc4f16ab4c112b11a3628e0ffcfb003e1200 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -174,6 +174,7 @@ Call::Call(const Call::Config& config) |
received_audio_bytes_per_sec_(1000, 1), |
received_rtcp_bytes_per_sec_(1000, 1), |
first_rtp_packet_received_ms_(-1) { |
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0); |
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps, |
config.bitrate_config.min_bitrate_bps); |
@@ -199,8 +200,8 @@ Call::Call(const Call::Config& config) |
} |
Call::~Call() { |
- UpdateHistograms(); |
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
+ UpdateHistograms(); |
RTC_CHECK(audio_send_ssrcs_.empty()); |
RTC_CHECK(video_send_ssrcs_.empty()); |
RTC_CHECK(video_send_streams_.empty()); |