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Unified Diff: webrtc/voice_engine/channel.cc

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updated RTP/RTCP module to use setter methods instead of passing the event log pointer in the const… Created 4 years, 10 months ago
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Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 9e27ce8b7b7162905a036f6d52a639c96d17bfa0..b661517bd77d3991663c8b22f8fd1caab1b974cc 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -19,6 +19,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/timeutils.h"
+#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
@@ -494,10 +495,13 @@ bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
}
int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
- if (event_log_) {
- unsigned int ssrc;
- RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
- event_log_->LogAudioPlayout(ssrc);
+ {
+ rtc::CritScope lock(&event_log_lock_);
+ if (event_log_) {
+ unsigned int ssrc;
+ RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+ { event_log_->LogAudioPlayout(ssrc); }
the sun 2016/03/03 09:25:13 Remove the { } on this line.
ivoc 2016/03/10 13:15:36 Hmm, not sure what happened there :-)
+ }
}
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame) ==
@@ -674,13 +678,12 @@ int32_t Channel::NeededFrequency(int32_t id) const {
int32_t Channel::CreateChannel(Channel*& channel,
int32_t channelId,
uint32_t instanceId,
- RtcEventLog* const event_log,
const Config& config) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
instanceId);
- channel = new Channel(channelId, instanceId, event_log, config);
+ channel = new Channel(channelId, instanceId, config);
if (channel == NULL) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel() unable to allocate memory for"
@@ -737,13 +740,10 @@ void Channel::RecordFileEnded(int32_t id) {
" shutdown");
}
-Channel::Channel(int32_t channelId,
- uint32_t instanceId,
- RtcEventLog* const event_log,
- const Config& config)
+Channel::Channel(int32_t channelId, uint32_t instanceId, const Config& config)
: _instanceId(instanceId),
_channelId(channelId),
- event_log_(event_log),
+ event_log_(nullptr),
rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
@@ -852,7 +852,6 @@ Channel::Channel(int32_t channelId,
seq_num_allocator_proxy_.get();
configuration.transport_feedback_callback = feedback_observer_proxy_.get();
}
- configuration.event_log = event_log;
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));

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