WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf
Convert audio/ and collateral (audio encoder copy red).
Bug: webrtc:10335
Change-Id: Iac54c0cfd2f62f4402f3deec35ae2725ec35b81a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36229}
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index b31ab04..dcafe3b 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -49,6 +49,7 @@
"../api:scoped_refptr",
"../api:sequence_checker",
"../api:transport_api",
+ "../api:webrtc_key_value_config",
"../api/audio:aec3_factory",
"../api/audio:audio_frame_api",
"../api/audio:audio_frame_processor",
@@ -190,6 +191,7 @@
"../test:mock_transformable_frame",
"../test:mock_transport",
"../test:rtp_test_utils",
+ "../test:scoped_key_value_config",
"../test:test_common",
"../test:test_support",
"utility:utility_tests",
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 08bb4e6..6aa0ec9 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -39,7 +39,6 @@
#include "rtc_base/logging.h"
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/task_queue.h"
-#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
@@ -88,8 +87,9 @@
"rate_prio", &bitrate_priority);
}
-AudioAllocationConfig::AudioAllocationConfig() {
- Parser()->Parse(field_trial::FindFullName(kKey));
+AudioAllocationConfig::AudioAllocationConfig(
+ const WebRtcKeyValueConfig& field_trials) {
+ Parser()->Parse(field_trials.Lookup(kKey));
if (priority_bitrate_raw && !priority_bitrate.IsZero()) {
RTC_LOG(LS_WARNING) << "'priority_bitrate' and '_raw' are mutually "
"exclusive but both were configured.";
@@ -106,28 +106,31 @@
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
- const absl::optional<RtpState>& suspended_rtp_state)
- : AudioSendStream(clock,
- config,
- audio_state,
- task_queue_factory,
- rtp_transport,
- bitrate_allocator,
- event_log,
- suspended_rtp_state,
- voe::CreateChannelSend(
- clock,
- task_queue_factory,
- config.send_transport,
- rtcp_rtt_stats,
- event_log,
- config.frame_encryptor,
- config.crypto_options,
- config.rtp.extmap_allow_mixed,
- config.rtcp_report_interval_ms,
- config.rtp.ssrc,
- config.frame_transformer,
- rtp_transport->transport_feedback_observer())) {}
+ const absl::optional<RtpState>& suspended_rtp_state,
+ const WebRtcKeyValueConfig& field_trials)
+ : AudioSendStream(
+ clock,
+ config,
+ audio_state,
+ task_queue_factory,
+ rtp_transport,
+ bitrate_allocator,
+ event_log,
+ suspended_rtp_state,
+ voe::CreateChannelSend(clock,
+ task_queue_factory,
+ config.send_transport,
+ rtcp_rtt_stats,
+ event_log,
+ config.frame_encryptor,
+ config.crypto_options,
+ config.rtp.extmap_allow_mixed,
+ config.rtcp_report_interval_ms,
+ config.rtp.ssrc,
+ config.frame_transformer,
+ rtp_transport->transport_feedback_observer(),
+ field_trials),
+ field_trials) {}
AudioSendStream::AudioSendStream(
Clock* clock,
@@ -138,21 +141,24 @@
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
const absl::optional<RtpState>& suspended_rtp_state,
- std::unique_ptr<voe::ChannelSendInterface> channel_send)
+ std::unique_ptr<voe::ChannelSendInterface> channel_send,
+ const WebRtcKeyValueConfig& field_trials)
: clock_(clock),
+ field_trials_(field_trials),
rtp_transport_queue_(rtp_transport->GetWorkerQueue()),
allocate_audio_without_feedback_(
- field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")),
+ field_trials_.IsEnabled("WebRTC-Audio-ABWENoTWCC")),
enable_audio_alr_probing_(
- !field_trial::IsDisabled("WebRTC-Audio-AlrProbing")),
+ !field_trials_.IsDisabled("WebRTC-Audio-AlrProbing")),
send_side_bwe_with_overhead_(
- !field_trial::IsDisabled("WebRTC-SendSideBwe-WithOverhead")),
+ !field_trials_.IsDisabled("WebRTC-SendSideBwe-WithOverhead")),
+ allocation_settings_(field_trials_),
config_(Config(/*send_transport=*/nullptr)),
audio_state_(audio_state),
channel_send_(std::move(channel_send)),
event_log_(event_log),
use_legacy_overhead_calculation_(
- field_trial::IsEnabled("WebRTC-Audio-LegacyOverhead")),
+ field_trials_.IsEnabled("WebRTC-Audio-LegacyOverhead")),
bitrate_allocator_(bitrate_allocator),
rtp_transport_(rtp_transport),
rtp_rtcp_module_(channel_send_->GetRtpRtcp()),
@@ -640,7 +646,8 @@
AudioEncoderCopyRed::Config red_config;
red_config.payload_type = *spec.red_payload_type;
red_config.speech_encoder = std::move(encoder);
- encoder = std::make_unique<AudioEncoderCopyRed>(std::move(red_config));
+ encoder = std::make_unique<AudioEncoderCopyRed>(std::move(red_config),
+ field_trials_);
}
// Set currently known overhead (used in ANA, opus only).
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index b407508..ad40a89 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -16,6 +16,7 @@
#include <vector>
#include "api/sequence_checker.h"
+#include "api/webrtc_key_value_config.h"
#include "audio/audio_level.h"
#include "audio/channel_send.h"
#include "call/audio_send_stream.h"
@@ -46,7 +47,7 @@
absl::optional<double> bitrate_priority;
std::unique_ptr<StructParametersParser> Parser();
- AudioAllocationConfig();
+ explicit AudioAllocationConfig(const WebRtcKeyValueConfig& field_trials);
};
namespace internal {
class AudioState;
@@ -62,7 +63,8 @@
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
- const absl::optional<RtpState>& suspended_rtp_state);
+ const absl::optional<RtpState>& suspended_rtp_state,
+ const WebRtcKeyValueConfig& field_trials);
// For unit tests, which need to supply a mock ChannelSend.
AudioSendStream(Clock* clock,
const webrtc::AudioSendStream::Config& config,
@@ -72,7 +74,8 @@
BitrateAllocatorInterface* bitrate_allocator,
RtcEventLog* event_log,
const absl::optional<RtpState>& suspended_rtp_state,
- std::unique_ptr<voe::ChannelSendInterface> channel_send);
+ std::unique_ptr<voe::ChannelSendInterface> channel_send,
+ const WebRtcKeyValueConfig& field_trials);
AudioSendStream() = delete;
AudioSendStream(const AudioSendStream&) = delete;
@@ -160,6 +163,7 @@
RTC_RUN_ON(worker_thread_checker_);
Clock* clock_;
+ const WebRtcKeyValueConfig& field_trials_;
SequenceChecker worker_thread_checker_;
SequenceChecker pacer_thread_checker_;
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 9228611..2d2e64c 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -32,10 +32,10 @@
#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "rtc_base/task_queue_for_test.h"
#include "system_wrappers/include/clock.h"
-#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
#include "test/mock_audio_encoder_factory.h"
+#include "test/scoped_key_value_config.h"
namespace webrtc {
namespace test {
@@ -196,7 +196,8 @@
Clock::GetRealTimeClock(), stream_config_, audio_state_,
task_queue_factory_.get(), &rtp_transport_, &bitrate_allocator_,
&event_log_, absl::nullopt,
- std::unique_ptr<voe::ChannelSendInterface>(channel_send_)));
+ std::unique_ptr<voe::ChannelSendInterface>(channel_send_),
+ field_trials));
}
AudioSendStream::Config& config() { return stream_config_; }
@@ -321,6 +322,8 @@
TaskQueueForTest* worker() { return &worker_queue_; }
+ test::ScopedKeyValueConfig field_trials;
+
private:
SimulatedClock clock_;
std::unique_ptr<TaskQueueFactory> task_queue_factory_;
@@ -659,10 +662,10 @@
}
TEST(AudioSendStreamTest, SSBweFieldTrialMinRespected) {
- ScopedFieldTrials field_trials(
- "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
+ ScopedKeyValueConfig field_trials(
+ helper.field_trials, "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(
*helper.channel_send(),
@@ -676,10 +679,10 @@
}
TEST(AudioSendStreamTest, SSBweFieldTrialMaxRespected) {
- ScopedFieldTrials field_trials(
- "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
+ ScopedKeyValueConfig field_trials(
+ helper.field_trials, "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
auto send_stream = helper.CreateAudioSendStream();
EXPECT_CALL(
*helper.channel_send(),
@@ -693,10 +696,10 @@
}
TEST(AudioSendStreamTest, SSBweWithOverhead) {
- ScopedFieldTrials field_trials(
- "WebRTC-Audio-LegacyOverhead/Disabled/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
+ ScopedKeyValueConfig field_trials(helper.field_trials,
+ "WebRTC-Audio-LegacyOverhead/Disabled/");
EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
.WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
auto send_stream = helper.CreateAudioSendStream();
@@ -714,11 +717,12 @@
}
TEST(AudioSendStreamTest, SSBweWithOverheadMinRespected) {
- ScopedFieldTrials field_trials(
- "WebRTC-Audio-LegacyOverhead/Disabled/"
- "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
+ ScopedKeyValueConfig field_trials(
+ helper.field_trials,
+ "WebRTC-Audio-LegacyOverhead/Disabled/"
+ "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
.WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
auto send_stream = helper.CreateAudioSendStream();
@@ -734,11 +738,12 @@
}
TEST(AudioSendStreamTest, SSBweWithOverheadMaxRespected) {
- ScopedFieldTrials field_trials(
- "WebRTC-Audio-LegacyOverhead/Disabled/"
- "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
for (bool use_null_audio_processing : {false, true}) {
ConfigHelper helper(true, true, use_null_audio_processing);
+ ScopedKeyValueConfig field_trials(
+ helper.field_trials,
+ "WebRTC-Audio-LegacyOverhead/Disabled/"
+ "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
.WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
auto send_stream = helper.CreateAudioSendStream();
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index d1135e5..d6b5823 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -43,7 +43,6 @@
#include "rtc_base/task_queue.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/clock.h"
-#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
@@ -78,7 +77,8 @@
int rtcp_report_interval_ms,
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
- TransportFeedbackObserver* feedback_observer);
+ TransportFeedbackObserver* feedback_observer,
+ const WebRtcKeyValueConfig& field_trials);
~ChannelSend() override;
@@ -458,7 +458,8 @@
int rtcp_report_interval_ms,
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
- TransportFeedbackObserver* feedback_observer)
+ TransportFeedbackObserver* feedback_observer,
+ const WebRtcKeyValueConfig& field_trials)
: ssrc_(ssrc),
event_log_(rtc_event_log),
_timeStamp(0), // This is just an offset, RTP module will add it's own
@@ -477,7 +478,7 @@
"AudioEncoder",
TaskQueueFactory::Priority::NORMAL)),
fixing_timestamp_stall_(
- !field_trial::IsDisabled("WebRTC-Audio-FixTimestampStall")) {
+ field_trials.IsDisabled("WebRTC-Audio-FixTimestampStall")) {
audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config()));
RtpRtcpInterface::Configuration configuration;
@@ -948,12 +949,13 @@
int rtcp_report_interval_ms,
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
- TransportFeedbackObserver* feedback_observer) {
+ TransportFeedbackObserver* feedback_observer,
+ const WebRtcKeyValueConfig& field_trials) {
return std::make_unique<ChannelSend>(
clock, task_queue_factory, rtp_transport, rtcp_rtt_stats, rtc_event_log,
frame_encryptor, crypto_options, extmap_allow_mixed,
rtcp_report_interval_ms, ssrc, std::move(frame_transformer),
- feedback_observer);
+ feedback_observer, field_trials);
}
} // namespace voe
diff --git a/audio/channel_send.h b/audio/channel_send.h
index e100725..bfbfbee 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -21,6 +21,7 @@
#include "api/frame_transformer_interface.h"
#include "api/function_view.h"
#include "api/task_queue/task_queue_factory.h"
+#include "api/webrtc_key_value_config.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
@@ -135,7 +136,8 @@
int rtcp_report_interval_ms,
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
- TransportFeedbackObserver* feedback_observer);
+ TransportFeedbackObserver* feedback_observer,
+ const WebRtcKeyValueConfig& field_trials);
} // namespace voe
} // namespace webrtc
diff --git a/call/call.cc b/call/call.cc
index 16f7b90..f2101cf 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -915,7 +915,7 @@
AudioSendStream* send_stream = new AudioSendStream(
clock_, config, config_.audio_state, task_queue_factory_,
transport_send_.get(), bitrate_allocator_.get(), event_log_,
- call_stats_->AsRtcpRttStats(), suspended_rtp_state);
+ call_stats_->AsRtcpRttStats(), suspended_rtp_state, trials());
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
audio_send_ssrcs_.end());
audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index c4814ce..af76a11 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -118,12 +118,12 @@
deps = [
"../../api:array_view",
+ "../../api:webrtc_key_value_config",
"../../api/audio_codecs:audio_codecs_api",
"../../api/units:time_delta",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
- "../../system_wrappers:field_trial",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
@@ -2079,6 +2079,7 @@
"../../test:fileutils",
"../../test:rtc_expect_death",
"../../test:rtp_test_utils",
+ "../../test:scoped_key_value_config",
"../../test:test_common",
"../../test:test_support",
"codecs/opus/test",
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 9643c7b..c8a26e8 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -18,7 +18,6 @@
#include "rtc_base/byte_order.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
-#include "system_wrappers/include/field_trial.h"
namespace webrtc {
static constexpr const int kRedMaxPacketSize =
@@ -40,9 +39,10 @@
AudioEncoderCopyRed::Config::Config(Config&&) = default;
AudioEncoderCopyRed::Config::~Config() = default;
-size_t GetMaxRedundancyFromFieldTrial() {
+size_t GetMaxRedundancyFromFieldTrial(
+ const WebRtcKeyValueConfig& field_trials) {
const std::string red_trial =
- webrtc::field_trial::FindFullName("WebRTC-Audio-Red-For-Opus");
+ field_trials.Lookup("WebRTC-Audio-Red-For-Opus");
size_t redundancy = 0;
if (sscanf(red_trial.c_str(), "Enabled-%zu", &redundancy) != 1 ||
redundancy > 9) {
@@ -51,14 +51,17 @@
return redundancy;
}
-AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config)
+AudioEncoderCopyRed::AudioEncoderCopyRed(
+ Config&& config,
+ const WebRtcKeyValueConfig& field_trials)
: speech_encoder_(std::move(config.speech_encoder)),
primary_encoded_(0, kAudioMaxRtpPacketLen),
max_packet_length_(kAudioMaxRtpPacketLen),
red_payload_type_(config.payload_type) {
RTC_CHECK(speech_encoder_) << "Speech encoder not provided.";
- auto number_of_redundant_encodings = GetMaxRedundancyFromFieldTrial();
+ auto number_of_redundant_encodings =
+ GetMaxRedundancyFromFieldTrial(field_trials);
for (size_t i = 0; i < number_of_redundant_encodings; i++) {
std::pair<EncodedInfo, rtc::Buffer> redundant;
redundant.second.EnsureCapacity(kAudioMaxRtpPacketLen);
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index d163193..e7471b3 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -22,6 +22,7 @@
#include "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/units/time_delta.h"
+#include "api/webrtc_key_value_config.h"
#include "rtc_base/buffer.h"
namespace webrtc {
@@ -42,7 +43,8 @@
std::unique_ptr<AudioEncoder> speech_encoder;
};
- explicit AudioEncoderCopyRed(Config&& config);
+ AudioEncoderCopyRed(Config&& config,
+ const WebRtcKeyValueConfig& field_trials);
~AudioEncoderCopyRed() override;
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index 0eeac01..795a996 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -18,6 +18,7 @@
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
+#include "test/scoped_key_value_config.h"
#include "test/testsupport/rtc_expect_death.h"
using ::testing::_;
@@ -49,7 +50,7 @@
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type_;
config.speech_encoder = std::unique_ptr<AudioEncoder>(mock_encoder_);
- red_.reset(new AudioEncoderCopyRed(std::move(config)));
+ red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials_));
memset(audio_, 0, sizeof(audio_));
EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(1U));
EXPECT_CALL(*mock_encoder_, SampleRateHz())
@@ -68,6 +69,7 @@
timestamp_ += rtc::checked_cast<uint32_t>(num_audio_samples_10ms);
}
+ test::ScopedKeyValueConfig field_trials_;
MockAudioEncoder* mock_encoder_;
std::unique_ptr<AudioEncoderCopyRed> red_;
uint32_t timestamp_;
@@ -198,13 +200,13 @@
// Checks that the correct payload sizes are populated into the redundancy
// information for a redundancy level of 0.
TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes0) {
- webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-Red-For-Opus/Enabled-0/");
+ webrtc::test::ScopedKeyValueConfig field_trials(
+ field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-0/");
// Recreate the RED encoder to take the new field trial setting into account.
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type_;
config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
- red_.reset(new AudioEncoderCopyRed(std::move(config)));
+ red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
// Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
// of calls.
@@ -224,13 +226,13 @@
// Checks that the correct payload sizes are populated into the redundancy
// information for a redundancy level of 2.
TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes2) {
- webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-Red-For-Opus/Enabled-2/");
+ webrtc::test::ScopedKeyValueConfig field_trials(
+ field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-2/");
// Recreate the RED encoder to take the new field trial setting into account.
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type_;
config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
- red_.reset(new AudioEncoderCopyRed(std::move(config)));
+ red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
// Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
// of calls.
@@ -266,13 +268,13 @@
// Checks that the correct payload sizes are populated into the redundancy
// information for a redundancy level of 3.
TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes3) {
- webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-Red-For-Opus/Enabled-3/");
+ webrtc::test::ScopedKeyValueConfig field_trials(
+ field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-3/");
// Recreate the RED encoder to take the new field trial setting into account.
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type_;
config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
- red_.reset(new AudioEncoderCopyRed(std::move(config)));
+ red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials_));
// Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
// of calls.
@@ -463,13 +465,13 @@
// Variant with a redundancy of 0.
TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header0) {
- webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-Red-For-Opus/Enabled-0/");
+ webrtc::test::ScopedKeyValueConfig field_trials(
+ field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-0/");
// Recreate the RED encoder to take the new field trial setting into account.
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type_;
config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
- red_.reset(new AudioEncoderCopyRed(std::move(config)));
+ red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
const int primary_payload_type = red_payload_type_ + 1;
AudioEncoder::EncodedInfo info;
@@ -491,13 +493,13 @@
}
// Variant with a redundancy of 2.
TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header2) {
- webrtc::test::ScopedFieldTrials field_trials(
- "WebRTC-Audio-Red-For-Opus/Enabled-2/");
+ webrtc::test::ScopedKeyValueConfig field_trials(
+ field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-2/");
// Recreate the RED encoder to take the new field trial setting into account.
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type_;
config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
- red_.reset(new AudioEncoderCopyRed(std::move(config)));
+ red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
const int primary_payload_type = red_payload_type_ + 1;
AudioEncoder::EncodedInfo info;
@@ -623,11 +625,13 @@
}
TEST_F(AudioEncoderCopyRedDeathTest, NullSpeechEncoder) {
+ test::ScopedKeyValueConfig field_trials;
AudioEncoderCopyRed* red = NULL;
AudioEncoderCopyRed::Config config;
config.speech_encoder = NULL;
- RTC_EXPECT_DEATH(red = new AudioEncoderCopyRed(std::move(config)),
- "Speech encoder not provided.");
+ RTC_EXPECT_DEATH(
+ red = new AudioEncoderCopyRed(std::move(config), field_trials),
+ "Speech encoder not provided.");
// The delete operation is needed to avoid leak reports from memcheck.
delete red;
}
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index d2c8d8a..892fbc8 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -190,7 +190,8 @@
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type;
config.speech_encoder = std::move(encoder);
- encoder = std::make_unique<AudioEncoderCopyRed>(std::move(config));
+ encoder = std::make_unique<AudioEncoderCopyRed>(std::move(config),
+ field_trials_);
receive_codecs.emplace(
std::make_pair(red_payload_type,
SdpAudioFormat("red", codec_format.clockrate_hz, 1)));
diff --git a/modules/audio_coding/test/TestRedFec.h b/modules/audio_coding/test/TestRedFec.h
index 0e92d27..dbadd88 100644
--- a/modules/audio_coding/test/TestRedFec.h
+++ b/modules/audio_coding/test/TestRedFec.h
@@ -19,6 +19,7 @@
#include "common_audio/vad/include/vad.h"
#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
+#include "test/scoped_key_value_config.h"
namespace webrtc {
@@ -37,6 +38,7 @@
void Run();
void OpenOutFile(int16_t testNumber);
+ test::ScopedKeyValueConfig field_trials_;
const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
std::unique_ptr<AudioCodingModule> _acmA;