| 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());
|
|
|