Refactor RtcEventAudioNetworkAdaptation to store config fields directly.
The event encoders and the event log writer are updated to use the new accessor methods.
Bug: None
Change-Id: I68be6be20337eac3e5e39c7095586ebd56380829
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/417781
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45961}
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
index fc42907..c0cba8d 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
@@ -45,7 +45,6 @@
#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
-#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
@@ -410,21 +409,20 @@
auto* audio_network_adaptation =
rtclog_event.mutable_audio_network_adaptation();
- if (event.config().bitrate_bps)
- audio_network_adaptation->set_bitrate_bps(*event.config().bitrate_bps);
- if (event.config().frame_length_ms)
- audio_network_adaptation->set_frame_length_ms(
- *event.config().frame_length_ms);
- if (event.config().uplink_packet_loss_fraction) {
+ if (event.bitrate_bps())
+ audio_network_adaptation->set_bitrate_bps(*event.bitrate_bps());
+ if (event.frame_length_ms())
+ audio_network_adaptation->set_frame_length_ms(*event.frame_length_ms());
+ if (event.uplink_packet_loss_fraction()) {
audio_network_adaptation->set_uplink_packet_loss_fraction(
- *event.config().uplink_packet_loss_fraction);
+ *event.uplink_packet_loss_fraction());
}
- if (event.config().enable_fec)
- audio_network_adaptation->set_enable_fec(*event.config().enable_fec);
- if (event.config().enable_dtx)
- audio_network_adaptation->set_enable_dtx(*event.config().enable_dtx);
- if (event.config().num_channels)
- audio_network_adaptation->set_num_channels(*event.config().num_channels);
+ if (event.enable_fec())
+ audio_network_adaptation->set_enable_fec(*event.enable_fec());
+ if (event.enable_dtx())
+ audio_network_adaptation->set_enable_dtx(*event.enable_dtx());
+ if (event.num_channels())
+ audio_network_adaptation->set_num_channels(*event.num_channels());
return Serialize(&rtclog_event);
}
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
index ab7297e..762c650 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
@@ -61,7 +61,6 @@
#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
-#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
@@ -972,28 +971,27 @@
rtclog2::AudioNetworkAdaptations* proto_batch =
event_stream->add_audio_network_adaptations();
proto_batch->set_timestamp_ms(base_event->timestamp_ms());
- if (base_event->config().bitrate_bps.has_value())
- proto_batch->set_bitrate_bps(base_event->config().bitrate_bps.value());
- if (base_event->config().frame_length_ms.has_value()) {
- proto_batch->set_frame_length_ms(
- base_event->config().frame_length_ms.value());
+ if (base_event->bitrate_bps().has_value())
+ proto_batch->set_bitrate_bps(base_event->bitrate_bps().value());
+ if (base_event->frame_length_ms().has_value()) {
+ proto_batch->set_frame_length_ms(base_event->frame_length_ms().value());
}
std::optional<uint64_t> base_uplink_packet_loss_fraction;
- if (base_event->config().uplink_packet_loss_fraction.has_value()) {
+ if (base_event->uplink_packet_loss_fraction().has_value()) {
base_uplink_packet_loss_fraction = ConvertPacketLossFractionToProtoFormat(
- base_event->config().uplink_packet_loss_fraction.value());
+ base_event->uplink_packet_loss_fraction().value());
proto_batch->set_uplink_packet_loss_fraction(
base_uplink_packet_loss_fraction.value());
}
- if (base_event->config().enable_fec.has_value())
- proto_batch->set_enable_fec(base_event->config().enable_fec.value());
- if (base_event->config().enable_dtx.has_value())
- proto_batch->set_enable_dtx(base_event->config().enable_dtx.value());
+ if (base_event->enable_fec().has_value())
+ proto_batch->set_enable_fec(base_event->enable_fec().value());
+ if (base_event->enable_dtx().has_value())
+ proto_batch->set_enable_dtx(base_event->enable_dtx().value());
// Note that `num_channels_deltas` encodes N as N-1, to keep deltas smaller,
// but there's no reason to do the same for the base event's value, since
// no bits will be spared.
- if (base_event->config().num_channels.has_value())
- proto_batch->set_num_channels(base_event->config().num_channels.value());
+ if (base_event->num_channels().has_value())
+ proto_batch->set_num_channels(base_event->num_channels().value());
if (batch.size() == 1)
return;
@@ -1016,15 +1014,15 @@
// bitrate_bps
for (size_t i = 0; i < values.size(); ++i) {
const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
- if (event->config().bitrate_bps.has_value()) {
- values[i] = ToUnsigned(event->config().bitrate_bps.value());
+ if (event->bitrate_bps().has_value()) {
+ values[i] = ToUnsigned(event->bitrate_bps().value());
} else {
values[i].reset();
}
}
const std::optional<uint64_t> unsigned_base_bitrate_bps =
- base_event->config().bitrate_bps.has_value()
- ? ToUnsigned(base_event->config().bitrate_bps.value())
+ base_event->bitrate_bps().has_value()
+ ? ToUnsigned(base_event->bitrate_bps().value())
: std::optional<uint64_t>();
encoded_deltas = EncodeDeltas(unsigned_base_bitrate_bps, values);
if (!encoded_deltas.empty()) {
@@ -1034,15 +1032,15 @@
// frame_length_ms
for (size_t i = 0; i < values.size(); ++i) {
const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
- if (event->config().frame_length_ms.has_value()) {
- values[i] = ToUnsigned(event->config().frame_length_ms.value());
+ if (event->frame_length_ms().has_value()) {
+ values[i] = ToUnsigned(event->frame_length_ms().value());
} else {
values[i].reset();
}
}
const std::optional<uint64_t> unsigned_base_frame_length_ms =
- base_event->config().frame_length_ms.has_value()
- ? ToUnsigned(base_event->config().frame_length_ms.value())
+ base_event->frame_length_ms().has_value()
+ ? ToUnsigned(base_event->frame_length_ms().value())
: std::optional<uint64_t>();
encoded_deltas = EncodeDeltas(unsigned_base_frame_length_ms, values);
if (!encoded_deltas.empty()) {
@@ -1052,9 +1050,9 @@
// uplink_packet_loss_fraction
for (size_t i = 0; i < values.size(); ++i) {
const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
- if (event->config().uplink_packet_loss_fraction.has_value()) {
+ if (event->uplink_packet_loss_fraction().has_value()) {
values[i] = ConvertPacketLossFractionToProtoFormat(
- event->config().uplink_packet_loss_fraction.value());
+ event->uplink_packet_loss_fraction().value());
} else {
values[i].reset();
}
@@ -1067,9 +1065,9 @@
// enable_fec
for (size_t i = 0; i < values.size(); ++i) {
const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
- values[i] = event->config().enable_fec;
+ values[i] = event->enable_fec();
}
- encoded_deltas = EncodeDeltas(base_event->config().enable_fec, values);
+ encoded_deltas = EncodeDeltas(base_event->enable_fec(), values);
if (!encoded_deltas.empty()) {
proto_batch->set_enable_fec_deltas(encoded_deltas);
}
@@ -1077,9 +1075,9 @@
// enable_dtx
for (size_t i = 0; i < values.size(); ++i) {
const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
- values[i] = event->config().enable_dtx;
+ values[i] = event->enable_dtx();
}
- encoded_deltas = EncodeDeltas(base_event->config().enable_dtx, values);
+ encoded_deltas = EncodeDeltas(base_event->enable_dtx(), values);
if (!encoded_deltas.empty()) {
proto_batch->set_enable_dtx_deltas(encoded_deltas);
}
@@ -1087,7 +1085,7 @@
// num_channels
for (size_t i = 0; i < values.size(); ++i) {
const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
- const std::optional<size_t> num_channels = event->config().num_channels;
+ const std::optional<size_t> num_channels = event->num_channels();
if (num_channels.has_value()) {
// Since the number of channels is always greater than 0, we can encode
// N channels as N-1, thereby making sure that we get smaller deltas.
@@ -1103,9 +1101,9 @@
// In the base event, N channels encoded as N channels, but for delta
// compression purposes, also shifted down by 1.
std::optional<size_t> shifted_base_num_channels;
- if (base_event->config().num_channels.has_value()) {
- RTC_DCHECK_GT(base_event->config().num_channels.value(), 0u);
- shifted_base_num_channels = base_event->config().num_channels.value() - 1;
+ if (base_event->num_channels().has_value()) {
+ RTC_DCHECK_GT(base_event->num_channels().value(), 0u);
+ shifted_base_num_channels = base_event->num_channels().value() - 1;
}
encoded_deltas = EncodeDeltas(shifted_base_num_channels, values);
if (!encoded_deltas.empty()) {
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
index ac0059b..69bae30 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
@@ -315,12 +315,12 @@
event_count_);
for (size_t i = 0; i < event_count_; ++i) {
if (i == 0 || !force_repeated_fields_) {
- auto runtime_config = std::make_unique<AudioEncoderRuntimeConfig>();
+ AudioEncoderRuntimeConfig runtime_config;
const int bitrate_bps =
checked_cast<int>(prng_.Rand(0, std::numeric_limits<int32_t>::max()));
- runtime_config->bitrate_bps = bitrate_bps;
- events[i] = std::make_unique<RtcEventAudioNetworkAdaptation>(
- std::move(runtime_config));
+ runtime_config.bitrate_bps = bitrate_bps;
+ events[i] =
+ std::make_unique<RtcEventAudioNetworkAdaptation>(runtime_config);
} else {
events[i] = events[0]->Copy();
}
@@ -333,11 +333,11 @@
event_count_);
for (size_t i = 0; i < event_count_; ++i) {
if (i == 0 || !force_repeated_fields_) {
- auto runtime_config = std::make_unique<AudioEncoderRuntimeConfig>();
+ AudioEncoderRuntimeConfig runtime_config;
const int frame_length_ms = prng_.Rand(1, 1000);
- runtime_config->frame_length_ms = frame_length_ms;
- events[i] = std::make_unique<RtcEventAudioNetworkAdaptation>(
- std::move(runtime_config));
+ runtime_config.frame_length_ms = frame_length_ms;
+ events[i] =
+ std::make_unique<RtcEventAudioNetworkAdaptation>(runtime_config);
} else {
events[i] = events[0]->Copy();
}
@@ -352,10 +352,10 @@
if (i == 0 || !force_repeated_fields_) {
// To simplify the test, we just check powers of two.
const float plr = std::pow(0.5f, prng_.Rand(1, 8));
- auto runtime_config = std::make_unique<AudioEncoderRuntimeConfig>();
- runtime_config->uplink_packet_loss_fraction = plr;
- events[i] = std::make_unique<RtcEventAudioNetworkAdaptation>(
- std::move(runtime_config));
+ AudioEncoderRuntimeConfig runtime_config;
+ runtime_config.uplink_packet_loss_fraction = plr;
+ events[i] =
+ std::make_unique<RtcEventAudioNetworkAdaptation>(runtime_config);
} else {
events[i] = events[0]->Copy();
}
@@ -368,10 +368,10 @@
event_count_);
for (size_t i = 0; i < event_count_; ++i) {
if (i == 0 || !force_repeated_fields_) {
- auto runtime_config = std::make_unique<AudioEncoderRuntimeConfig>();
- runtime_config->enable_fec = prng_.Rand<bool>();
- events[i] = std::make_unique<RtcEventAudioNetworkAdaptation>(
- std::move(runtime_config));
+ AudioEncoderRuntimeConfig runtime_config;
+ runtime_config.enable_fec = prng_.Rand<bool>();
+ events[i] =
+ std::make_unique<RtcEventAudioNetworkAdaptation>(runtime_config);
} else {
events[i] = events[0]->Copy();
}
@@ -384,8 +384,8 @@
event_count_);
for (size_t i = 0; i < event_count_; ++i) {
if (i == 0 || !force_repeated_fields_) {
- auto runtime_config = std::make_unique<AudioEncoderRuntimeConfig>();
- runtime_config->enable_dtx = prng_.Rand<bool>();
+ AudioEncoderRuntimeConfig runtime_config;
+ runtime_config.enable_dtx = prng_.Rand<bool>();
events[i] = std::make_unique<RtcEventAudioNetworkAdaptation>(
std::move(runtime_config));
} else {
@@ -400,10 +400,10 @@
event_count_);
for (size_t i = 0; i < event_count_; ++i) {
if (i == 0 || !force_repeated_fields_) {
- auto runtime_config = std::make_unique<AudioEncoderRuntimeConfig>();
- runtime_config->num_channels = prng_.Rand(1, 2);
- events[i] = std::make_unique<RtcEventAudioNetworkAdaptation>(
- std::move(runtime_config));
+ AudioEncoderRuntimeConfig runtime_config;
+ runtime_config.num_channels = prng_.Rand(1, 2);
+ events[i] =
+ std::make_unique<RtcEventAudioNetworkAdaptation>(runtime_config);
} else {
events[i] = events[0]->Copy();
}
@@ -416,17 +416,17 @@
event_count_);
for (size_t i = 0; i < event_count_; ++i) {
if (i == 0 || !force_repeated_fields_) {
- auto runtime_config = std::make_unique<AudioEncoderRuntimeConfig>();
- runtime_config->bitrate_bps =
+ AudioEncoderRuntimeConfig runtime_config;
+ runtime_config.bitrate_bps =
checked_cast<int>(prng_.Rand(0, std::numeric_limits<int32_t>::max()));
- runtime_config->frame_length_ms = prng_.Rand(1, 1000);
- runtime_config->uplink_packet_loss_fraction =
+ runtime_config.frame_length_ms = prng_.Rand(1, 1000);
+ runtime_config.uplink_packet_loss_fraction =
std::pow(0.5f, prng_.Rand(1, 8));
- runtime_config->enable_fec = prng_.Rand<bool>();
- runtime_config->enable_dtx = prng_.Rand<bool>();
- runtime_config->num_channels = prng_.Rand(1, 2);
- events[i] = std::make_unique<RtcEventAudioNetworkAdaptation>(
- std::move(runtime_config));
+ runtime_config.enable_fec = prng_.Rand<bool>();
+ runtime_config.enable_dtx = prng_.Rand<bool>();
+ runtime_config.num_channels = prng_.Rand(1, 2);
+ events[i] =
+ std::make_unique<RtcEventAudioNetworkAdaptation>(runtime_config);
} else {
events[i] = events[0]->Copy();
}
diff --git a/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc b/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc
index 3265bf0..7bb1c27 100644
--- a/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc
+++ b/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc
@@ -11,25 +11,20 @@
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
#include <memory>
-#include <utility>
#include "absl/memory/memory.h"
-#include "api/rtc_event_log/rtc_event.h"
#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
-#include "rtc_base/checks.h"
namespace webrtc {
RtcEventAudioNetworkAdaptation::RtcEventAudioNetworkAdaptation(
- std::unique_ptr<AudioEncoderRuntimeConfig> config)
- : config_(std::move(config)) {
- RTC_DCHECK(config_);
-}
-
-RtcEventAudioNetworkAdaptation::RtcEventAudioNetworkAdaptation(
- const RtcEventAudioNetworkAdaptation& other)
- : RtcEvent(other),
- config_(std::make_unique<AudioEncoderRuntimeConfig>(*other.config_)) {}
+ const AudioEncoderRuntimeConfig& config)
+ : bitrate_bps_(config.bitrate_bps),
+ frame_length_ms_(config.frame_length_ms),
+ uplink_packet_loss_fraction_(config.uplink_packet_loss_fraction),
+ enable_fec_(config.enable_fec),
+ enable_dtx_(config.enable_dtx),
+ num_channels_(config.num_channels) {}
RtcEventAudioNetworkAdaptation::~RtcEventAudioNetworkAdaptation() = default;
diff --git a/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h b/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
index 7127a94..ea78ef4 100644
--- a/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
+++ b/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
@@ -11,8 +11,10 @@
#ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_NETWORK_ADAPTATION_H_
#define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_NETWORK_ADAPTATION_H_
+#include <cstddef>
#include <cstdint>
#include <memory>
+#include <optional>
#include <string>
#include <vector>
@@ -46,7 +48,8 @@
static constexpr Type kType = Type::AudioNetworkAdaptation;
explicit RtcEventAudioNetworkAdaptation(
- std::unique_ptr<AudioEncoderRuntimeConfig> config);
+ const AudioEncoderRuntimeConfig& config);
+
~RtcEventAudioNetworkAdaptation() override;
Type GetType() const override { return kType; }
@@ -54,7 +57,14 @@
std::unique_ptr<RtcEventAudioNetworkAdaptation> Copy() const;
- const AudioEncoderRuntimeConfig& config() const { return *config_; }
+ const std::optional<int>& bitrate_bps() const { return bitrate_bps_; }
+ const std::optional<int>& frame_length_ms() const { return frame_length_ms_; }
+ const std::optional<float>& uplink_packet_loss_fraction() const {
+ return uplink_packet_loss_fraction_;
+ }
+ const std::optional<bool>& enable_fec() const { return enable_fec_; }
+ const std::optional<bool>& enable_dtx() const { return enable_dtx_; }
+ const std::optional<size_t>& num_channels() const { return num_channels_; }
static std::string Encode(ArrayView<const RtcEvent*> /* batch */) {
// TODO(terelius): Implement
@@ -70,9 +80,15 @@
}
private:
- RtcEventAudioNetworkAdaptation(const RtcEventAudioNetworkAdaptation& other);
+ RtcEventAudioNetworkAdaptation(const RtcEventAudioNetworkAdaptation&) =
+ default;
- const std::unique_ptr<const AudioEncoderRuntimeConfig> config_;
+ std::optional<int> bitrate_bps_;
+ std::optional<int> frame_length_ms_;
+ std::optional<float> uplink_packet_loss_fraction_;
+ std::optional<bool> enable_fec_;
+ std::optional<bool> enable_dtx_;
+ std::optional<size_t> num_channels_;
};
} // namespace webrtc
diff --git a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
index 67f2007..9d05e0b 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
@@ -161,17 +161,16 @@
std::unique_ptr<RtcEventAudioNetworkAdaptation>
EventGenerator::NewAudioNetworkAdaptation() {
- std::unique_ptr<AudioEncoderRuntimeConfig> config =
- std::make_unique<AudioEncoderRuntimeConfig>();
+ AudioEncoderRuntimeConfig config;
- config->bitrate_bps = prng_.Rand(0, 3000000);
- config->enable_fec = prng_.Rand<bool>();
- config->enable_dtx = prng_.Rand<bool>();
- config->frame_length_ms = prng_.Rand(10, 120);
- config->num_channels = prng_.Rand(1, 2);
- config->uplink_packet_loss_fraction = prng_.Rand<float>();
+ config.bitrate_bps = prng_.Rand(0, 3000000);
+ config.enable_fec = prng_.Rand<bool>();
+ config.enable_dtx = prng_.Rand<bool>();
+ config.frame_length_ms = prng_.Rand(10, 120);
+ config.num_channels = prng_.Rand(1, 2);
+ config.uplink_packet_loss_fraction = prng_.Rand<float>();
- return std::make_unique<RtcEventAudioNetworkAdaptation>(std::move(config));
+ return std::make_unique<RtcEventAudioNetworkAdaptation>(config);
}
std::unique_ptr<RtcEventNetEqSetMinimumDelay>
@@ -787,21 +786,18 @@
const LoggedAudioNetworkAdaptationEvent& logged_event) const {
EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
- EXPECT_EQ(original_event.config().bitrate_bps,
- logged_event.config.bitrate_bps);
- EXPECT_EQ(original_event.config().enable_dtx, logged_event.config.enable_dtx);
- EXPECT_EQ(original_event.config().enable_fec, logged_event.config.enable_fec);
- EXPECT_EQ(original_event.config().frame_length_ms,
+ EXPECT_EQ(original_event.bitrate_bps(), logged_event.config.bitrate_bps);
+ EXPECT_EQ(original_event.enable_dtx(), logged_event.config.enable_dtx);
+ EXPECT_EQ(original_event.enable_fec(), logged_event.config.enable_fec);
+ EXPECT_EQ(original_event.frame_length_ms(),
logged_event.config.frame_length_ms);
- EXPECT_EQ(original_event.config().num_channels,
- logged_event.config.num_channels);
+ EXPECT_EQ(original_event.num_channels(), logged_event.config.num_channels);
// uplink_packet_loss_fraction
- ASSERT_EQ(original_event.config().uplink_packet_loss_fraction.has_value(),
+ ASSERT_EQ(original_event.uplink_packet_loss_fraction().has_value(),
logged_event.config.uplink_packet_loss_fraction.has_value());
- if (original_event.config().uplink_packet_loss_fraction.has_value()) {
- const float original =
- original_event.config().uplink_packet_loss_fraction.value();
+ if (original_event.uplink_packet_loss_fraction().has_value()) {
+ const float original = original_event.uplink_packet_loss_fraction().value();
const float logged =
logged_event.config.uplink_packet_loss_fraction.value();
const float uplink_packet_loss_fraction_delta = std::abs(original - logged);
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index dd8c9fe..8388cf4 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -55,7 +55,13 @@
return false;
}
auto ana_event = static_cast<RtcEventAudioNetworkAdaptation*>(arg);
- return ana_event->config() == config;
+ return ana_event->bitrate_bps() == config.bitrate_bps &&
+ ana_event->frame_length_ms() == config.frame_length_ms &&
+ ana_event->uplink_packet_loss_fraction() ==
+ config.uplink_packet_loss_fraction &&
+ ana_event->enable_fec() == config.enable_fec &&
+ ana_event->enable_dtx() == config.enable_dtx &&
+ ana_event->num_channels() == config.num_channels;
}
MATCHER_P(EncoderRuntimeConfigIs, config, "") {
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
index e9bffa2..6ab06c5 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
+++ b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
@@ -15,7 +15,6 @@
#include <cstdlib>
#include <memory>
#include <optional>
-#include <utility>
#include "api/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
@@ -67,9 +66,7 @@
}
void EventLogWriter::LogEncoderConfig(const AudioEncoderRuntimeConfig& config) {
- auto config_copy = std::make_unique<AudioEncoderRuntimeConfig>(config);
- event_log_->Log(
- std::make_unique<RtcEventAudioNetworkAdaptation>(std::move(config_copy)));
+ event_log_->Log(std::make_unique<RtcEventAudioNetworkAdaptation>(config));
last_logged_config_ = config;
}
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
index 1ae5794..becbcbe 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
@@ -41,7 +41,13 @@
return false;
}
auto ana_event = static_cast<RtcEventAudioNetworkAdaptation*>(arg);
- return ana_event->config() == config;
+ return ana_event->bitrate_bps() == config.bitrate_bps &&
+ ana_event->frame_length_ms() == config.frame_length_ms &&
+ ana_event->uplink_packet_loss_fraction() ==
+ config.uplink_packet_loss_fraction &&
+ ana_event->enable_fec() == config.enable_fec &&
+ ana_event->enable_dtx() == config.enable_dtx &&
+ ana_event->num_channels() == config.num_channels;
}
struct EventLogWriterStates {
