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WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf

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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}
This commit is contained in:
Jonas Oreland
2022-03-16 13:50:58 +01:00
committed by WebRTC LUCI CQ
parent e72f4bc524
commit a943e730b2
13 changed files with 117 additions and 82 deletions
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33
audio/audio_send_stream_unittest.cc
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@ -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 @@ struct ConfigHelper {
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 @@ struct ConfigHelper {
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, SSBweTargetInRangeRespected) {
}
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, SSBweFieldTrialMinRespected) {
}
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, SSBweFieldTrialMaxRespected) {
}
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, SSBweWithOverhead) {
}
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, SSBweWithOverheadMinRespected) {
}
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();