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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: Another rebase and accompanying changes. Created 4 years, 6 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 943a04572be088689cb655578001dd533703c0b7..8287668258986fe10a57738fc6cc9cd9a93c142c 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_coding/codecs/builtin_audio_decoder_factory.h"
@@ -58,6 +59,87 @@ bool RegisterReceiveCodec(std::unique_ptr<AudioCodingModule>* acm,
const int kTelephoneEventAttenuationdB = 10;
+class RtcEventLogProxy final : public webrtc::RtcEventLog {
+ public:
+ RtcEventLogProxy() : event_log_(nullptr) {}
+
+ bool StartLogging(const std::string& file_name,
+ int64_t max_size_bytes) override {
+ RTC_NOTREACHED();
+ return false;
+ }
+
+ bool StartLogging(rtc::PlatformFile log_file,
+ int64_t max_size_bytes) override {
+ RTC_NOTREACHED();
+ return false;
+ }
+
+ void StopLogging() override { RTC_NOTREACHED(); }
+
+ void LogVideoReceiveStreamConfig(
+ const webrtc::VideoReceiveStream::Config& config) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogVideoReceiveStreamConfig(config);
+ }
+ }
+
+ void LogVideoSendStreamConfig(
+ const webrtc::VideoSendStream::Config& config) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogVideoSendStreamConfig(config);
+ }
+ }
+
+ void LogRtpHeader(webrtc::PacketDirection direction,
+ webrtc::MediaType media_type,
+ const uint8_t* header,
+ size_t packet_length) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogRtpHeader(direction, media_type, header, packet_length);
+ }
+ }
+
+ void LogRtcpPacket(webrtc::PacketDirection direction,
+ webrtc::MediaType media_type,
+ const uint8_t* packet,
+ size_t length) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogRtcpPacket(direction, media_type, packet, length);
+ }
+ }
+
+ void LogAudioPlayout(uint32_t ssrc) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogAudioPlayout(ssrc);
+ }
+ }
+
+ void LogBwePacketLossEvent(int32_t bitrate,
+ uint8_t fraction_loss,
+ int32_t total_packets) override {
+ rtc::CritScope lock(&crit_);
+ if (event_log_) {
+ event_log_->LogBwePacketLossEvent(bitrate, fraction_loss, total_packets);
+ }
+ }
+
+ void SetEventLog(RtcEventLog* event_log) {
+ rtc::CritScope lock(&crit_);
+ event_log_ = event_log;
+ }
+
+ private:
+ rtc::CriticalSection crit_;
+ RtcEventLog* event_log_ GUARDED_BY(crit_);
+ RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogProxy);
+};
+
class TransportFeedbackProxy : public TransportFeedbackObserver {
public:
TransportFeedbackProxy() : feedback_observer_(nullptr) {
@@ -480,11 +562,9 @@ bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
int32_t id,
AudioFrame* audioFrame) {
- if (event_log_) {
- unsigned int ssrc;
- RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
- event_log_->LogAudioPlayout(ssrc);
- }
+ unsigned int ssrc;
+ RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+ event_log_proxy_->LogAudioPlayout(ssrc);
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
bool muted;
if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
@@ -671,9 +751,8 @@ 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) {
- return CreateChannel(channel, channelId, instanceId, event_log, config,
+ return CreateChannel(channel, channelId, instanceId, config,
CreateBuiltinAudioDecoderFactory());
}
@@ -681,15 +760,13 @@ int32_t Channel::CreateChannel(
Channel*& channel,
int32_t channelId,
uint32_t instanceId,
- RtcEventLog* const event_log,
const Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
instanceId);
- channel =
- new Channel(channelId, instanceId, event_log, config, decoder_factory);
+ channel = new Channel(channelId, instanceId, config, decoder_factory);
if (channel == NULL) {
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
"Channel::CreateChannel() unable to allocate memory for"
@@ -748,12 +825,11 @@ void Channel::RecordFileEnded(int32_t id) {
Channel::Channel(int32_t channelId,
uint32_t instanceId,
- RtcEventLog* const event_log,
const Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
: _instanceId(instanceId),
_channelId(channelId),
- event_log_(event_log),
+ event_log_proxy_(new RtcEventLogProxy()),
rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
@@ -856,7 +932,7 @@ Channel::Channel(int32_t channelId,
seq_num_allocator_proxy_.get();
configuration.transport_feedback_callback = feedback_observer_proxy_.get();
}
- configuration.event_log = event_log;
+ configuration.event_log = &(*event_log_proxy_);
_rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
_rtpRtcpModule->SetSendingMediaStatus(false);
@@ -3008,6 +3084,10 @@ void Channel::DisassociateSendChannel(int channel_id) {
}
}
+void Channel::SetRtcEventLog(RtcEventLog* event_log) {
+ event_log_proxy_->SetEventLog(event_log);
+}
+
int Channel::RegisterExternalMediaProcessing(ProcessingTypes type,
VoEMediaProcess& processObject) {
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
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