Index: webrtc/call.h |
diff --git a/webrtc/call.h b/webrtc/call.h |
index 193124ee179879207b7218d26a6d46ee5a48cbd8..9855b600f050ebd64cd02a5639869c0b59c4e38e 100644 |
--- a/webrtc/call.h |
+++ b/webrtc/call.h |
@@ -26,6 +26,7 @@ |
namespace webrtc { |
class AudioProcessing; |
+class RtcEventLog; |
const char* Version(); |
@@ -72,6 +73,10 @@ class LoadObserver { |
class Call { |
public: |
struct Config { |
+ explicit Config(RtcEventLog* event_log) : event_log(event_log) { |
+ RTC_DCHECK(event_log); |
+ } |
+ |
static const int kDefaultStartBitrateBps; |
// Bitrate config used until valid bitrate estimates are calculated. Also |
@@ -89,6 +94,10 @@ class Call { |
// Audio Processing Module to be used in this call. |
// TODO(solenberg): Change this to a shared_ptr once we can use C++11. |
AudioProcessing* audio_processing = nullptr; |
+ |
+ // RtcEventLog to use for this call. Required. |
+ // Use webrtc::RtcEventLog::CreateNull() for a null implementation. |
+ RtcEventLog* event_log = nullptr; |
}; |
struct Stats { |
@@ -151,10 +160,6 @@ class Call { |
virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; |
- virtual bool StartEventLog(rtc::PlatformFile log_file, |
- int64_t max_size_bytes) = 0; |
- virtual void StopEventLog() = 0; |
- |
virtual ~Call() {} |
}; |