Added support for logging the SSRC corresponding to AudioPlayout events.
To do this, the logging of this event was moved from the ACM to
VoiceEngine Channel. A new LogAudioPlayoutEvent function was added on
the RtcEventLog interface, and the LogDebugEvent function was removed
since it is no longer being used.
BUG=webrtc:4741
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, kwiberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/1340283002 .
Cr-Commit-Position: refs/heads/master@{#9972}
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 3013925..98b5265 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -27,7 +27,6 @@
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/typedefs.h"
-#include "webrtc/video/rtc_event_log.h"
namespace webrtc {
@@ -147,8 +146,7 @@
first_frame_(true),
callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
packetization_callback_(NULL),
- vad_callback_(NULL),
- event_log_(config.event_log) {
+ vad_callback_(NULL) {
if (InitializeReceiverSafe() < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot initialize receiver");
@@ -682,11 +680,6 @@
"PlayoutData failed, RecOut Failed");
return -1;
}
- {
- if (event_log_)
- event_log_->LogDebugEvent(RtcEventLog::DebugEvent::kAudioPlayout);
- }
-
audio_frame->id_ = id_;
return 0;
}
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 837cd11..f7bad88 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -295,8 +295,6 @@
AudioPacketizationCallback* packetization_callback_
GUARDED_BY(callback_crit_sect_);
ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
-
- RtcEventLog* const event_log_;
};
} // namespace acm2
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index 0d3d5da..9ce622f 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -29,7 +29,6 @@
class AudioDecoder;
class AudioEncoder;
class AudioFrame;
-class RtcEventLog;
class RTPFragmentationHeader;
#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
@@ -83,16 +82,11 @@
public:
struct Config {
- Config()
- : id(0),
- neteq_config(),
- clock(Clock::GetRealTimeClock()),
- event_log(nullptr) {}
+ Config() : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {}
int id;
NetEq::Config neteq_config;
Clock* clock;
- RtcEventLog* event_log;
};
///////////////////////////////////////////////////////////////////////////
diff --git a/webrtc/video/rtc_event_log.cc b/webrtc/video/rtc_event_log.cc
index 7086b3e..8e6d1b9 100644
--- a/webrtc/video/rtc_event_log.cc
+++ b/webrtc/video/rtc_event_log.cc
@@ -50,7 +50,7 @@
MediaType media_type,
const uint8_t* packet,
size_t length) override {}
- void LogDebugEvent(DebugEvent event_type) override {}
+ void LogAudioPlayout(uint32_t ssrc) override {}
};
#else // ENABLE_RTC_EVENT_LOG is defined
@@ -72,7 +72,7 @@
MediaType media_type,
const uint8_t* packet,
size_t length) override;
- void LogDebugEvent(DebugEvent event_type) override;
+ void LogAudioPlayout(uint32_t ssrc) override;
private:
// Stops logging and clears the stored data and buffers.
@@ -111,20 +111,6 @@
// unnamed namespace. The intention is to make the compiler warn if anyone
// adds unhandled new events/modes/etc.
-rtclog::DebugEvent_EventType ConvertDebugEvent(
- RtcEventLog::DebugEvent event_type) {
- switch (event_type) {
- case RtcEventLog::DebugEvent::kLogStart:
- return rtclog::DebugEvent::LOG_START;
- case RtcEventLog::DebugEvent::kLogEnd:
- return rtclog::DebugEvent::LOG_END;
- case RtcEventLog::DebugEvent::kAudioPlayout:
- return rtclog::DebugEvent::AUDIO_PLAYOUT;
- }
- RTC_NOTREACHED();
- return rtclog::DebugEvent::UNKNOWN_EVENT;
-}
-
rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(
newapi::RtcpMode rtcp_mode) {
switch (rtcp_mode) {
@@ -188,7 +174,7 @@
start_event.set_timestamp_us(start_time_us_);
start_event.set_type(rtclog::Event::DEBUG_EVENT);
auto debug_event = start_event.mutable_debug_event();
- debug_event->set_type(ConvertDebugEvent(DebugEvent::kLogStart));
+ debug_event->set_type(rtclog::DebugEvent_EventType_LOG_START);
StoreToFile(&start_event);
}
@@ -330,14 +316,15 @@
HandleEvent(&rtcp_event);
}
-void RtcEventLogImpl::LogDebugEvent(DebugEvent event_type) {
+void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) {
rtc::CritScope lock(&crit_);
rtclog::Event event;
const int64_t timestamp = clock_->TimeInMicroseconds();
event.set_timestamp_us(timestamp);
event.set_type(rtclog::Event::DEBUG_EVENT);
auto debug_event = event.mutable_debug_event();
- debug_event->set_type(ConvertDebugEvent(event_type));
+ debug_event->set_type(rtclog::DebugEvent_EventType_AUDIO_PLAYOUT);
+ debug_event->set_local_ssrc(ssrc);
HandleEvent(&event);
}
@@ -350,7 +337,7 @@
event.set_timestamp_us(timestamp);
event.set_type(rtclog::Event::DEBUG_EVENT);
auto debug_event = event.mutable_debug_event();
- debug_event->set_type(ConvertDebugEvent(DebugEvent::kLogEnd));
+ debug_event->set_type(rtclog::DebugEvent_EventType_LOG_END);
// Store the event and close the file
RTC_DCHECK(file_->Open());
StoreToFile(&event);
diff --git a/webrtc/video/rtc_event_log.h b/webrtc/video/rtc_event_log.h
index a58d745..daab3c1 100644
--- a/webrtc/video/rtc_event_log.h
+++ b/webrtc/video/rtc_event_log.h
@@ -31,9 +31,6 @@
class RtcEventLog {
public:
- // The types of debug events that are currently supported for logging.
- enum class DebugEvent { kLogStart, kLogEnd, kAudioPlayout };
-
virtual ~RtcEventLog() {}
static rtc::scoped_ptr<RtcEventLog> Create();
@@ -67,8 +64,8 @@
const uint8_t* packet,
size_t length) = 0;
- // Logs a debug event.
- virtual void LogDebugEvent(DebugEvent event_type) = 0;
+ // Logs an audio playout event
+ virtual void LogAudioPlayout(uint32_t ssrc) = 0;
// Reads an RtcEventLog file and returns true when reading was successful.
// The result is stored in the given EventStream object.
diff --git a/webrtc/video/rtc_event_log.proto b/webrtc/video/rtc_event_log.proto
index 7e4e699..bdb6bb1 100644
--- a/webrtc/video/rtc_event_log.proto
+++ b/webrtc/video/rtc_event_log.proto
@@ -106,6 +106,9 @@
// required
optional EventType type = 1;
+
+ // required if type == AUDIO_PLAYOUT
+ optional uint32 local_ssrc = 2;
}
diff --git a/webrtc/video/rtc_event_log_unittest.cc b/webrtc/video/rtc_event_log_unittest.cc
index 7a2bd11..6c1786b 100644
--- a/webrtc/video/rtc_event_log_unittest.cc
+++ b/webrtc/video/rtc_event_log_unittest.cc
@@ -265,12 +265,14 @@
}
}
-void VerifyPlayoutEvent(const rtclog::Event& event) {
+void VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) {
ASSERT_TRUE(IsValidBasicEvent(event));
ASSERT_EQ(rtclog::Event::DEBUG_EVENT, event.type());
const rtclog::DebugEvent& debug_event = event.debug_event();
ASSERT_TRUE(debug_event.has_type());
EXPECT_EQ(rtclog::DebugEvent::AUDIO_PLAYOUT, debug_event.type());
+ ASSERT_TRUE(debug_event.has_local_ssrc());
+ EXPECT_EQ(ssrc, debug_event.local_ssrc());
}
void VerifyLogStartEvent(const rtclog::Event& event) {
@@ -407,6 +409,7 @@
std::vector<rtc::Buffer> rtp_packets;
std::vector<rtc::Buffer> rtcp_packets;
std::vector<size_t> rtp_header_sizes;
+ std::vector<uint32_t> playout_ssrcs;
VideoReceiveStream::Config receiver_config(nullptr);
VideoSendStream::Config sender_config(nullptr);
@@ -427,6 +430,10 @@
rtcp_packets.push_back(rtc::Buffer(packet_size));
GenerateRtcpPacket(rtcp_packets[i].data(), packet_size);
}
+ // Create debug_count random SSRCs to use when logging AudioPlayout events.
+ for (size_t i = 0; i < debug_count; i++) {
+ playout_ssrcs.push_back(static_cast<uint32_t>(rand()));
+ }
// Create configurations for the video streams.
GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config);
GenerateVideoSendConfig(extensions_bitvector, &sender_config);
@@ -459,7 +466,7 @@
rtcp_index++;
}
if (i * debug_count >= debug_index * rtp_count) {
- log_dumper->LogDebugEvent(RtcEventLog::DebugEvent::kAudioPlayout);
+ log_dumper->LogAudioPlayout(playout_ssrcs[debug_index - 1]);
debug_index++;
}
if (i == rtp_count / 2) {
@@ -497,7 +504,8 @@
rtcp_index++;
}
if (i * debug_count >= debug_index * rtp_count) {
- VerifyPlayoutEvent(parsed_stream.stream(event_index));
+ VerifyPlayoutEvent(parsed_stream.stream(event_index),
+ playout_ssrcs[debug_index - 1]);
event_index++;
debug_index++;
}
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index fb29dfa..a8fb179 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -12,6 +12,7 @@
#include <algorithm>
+#include "webrtc/base/checks.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/common.h"
@@ -450,6 +451,11 @@
WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::GetAudioFrame(id=%d)", id);
+ if (event_log_) {
+ unsigned int ssrc;
+ RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+ event_log_->LogAudioPlayout(ssrc);
+ }
// Get 10ms raw PCM data from the ACM (mixer limits output frequency)
if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_,
audioFrame) == -1)
@@ -719,6 +725,7 @@
volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
_instanceId(instanceId),
_channelId(channelId),
+ event_log_(event_log),
rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
@@ -809,7 +816,6 @@
}
acm_config.neteq_config.enable_fast_accelerate =
config.Get<NetEqFastAccelerate>().enabled;
- acm_config.event_log = event_log;
audio_coding_.reset(AudioCodingModule::Create(acm_config));
_inbandDtmfQueue.ResetDtmf();
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 56f6ce4..66eaff1 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -488,6 +488,8 @@
ChannelState channel_state_;
+ RtcEventLog* const event_log_;
+
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
