Adds trial to calculate audio overhead based on available data.
This adds the ability to disable legacy overhead calculation so we'll
use the available data on per packet over head and frame length range
to set the min and max total allocatable bitrate.
Bug: webrtc:11001
Change-Id: I2a94499433e15bad11a08f81fe7f1dfc27982cdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155175
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29368}
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 948fcfe..d787a8a 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -83,8 +83,10 @@
// should be made more precise in the future. This can be changed when that
// logic is more accurate.
const DataSize kOverheadPerPacket = DataSize::bytes(20 + 8 + 10 + 12);
-const TimeDelta kMaxFrameLength = TimeDelta::ms(60);
-const DataRate kOverheadRate = kOverheadPerPacket / kMaxFrameLength;
+const TimeDelta kMinFrameLength = TimeDelta::ms(20);
+const TimeDelta kMaxFrameLength = TimeDelta::ms(120);
+const DataRate kMinOverheadRate = kOverheadPerPacket / kMaxFrameLength;
+const DataRate kMaxOverheadRate = kOverheadPerPacket / kMinFrameLength;
class MockLimitObserver : public BitrateAllocator::LimitObserver {
public:
@@ -104,6 +106,9 @@
.WillByDefault(Return(spec.info.num_channels));
ON_CALL(*encoder.get(), RtpTimestampRateHz())
.WillByDefault(Return(spec.format.clockrate_hz));
+ ON_CALL(*encoder.get(), GetFrameLengthRange())
+ .WillByDefault(Return(absl::optional<std::pair<TimeDelta, TimeDelta>>{
+ {TimeDelta::ms(20), TimeDelta::ms(120)}}));
return encoder;
}
}
@@ -299,6 +304,7 @@
EXPECT_CALL(*audio_processing_, GetStatistics(true))
.WillRepeatedly(Return(audio_processing_stats_));
}
+ TaskQueueForTest* worker() { return &worker_queue_; }
private:
SimulatedClock clock_;
@@ -546,7 +552,7 @@
update.packet_loss_ratio = 0;
update.round_trip_time = TimeDelta::ms(50);
update.bwe_period = TimeDelta::ms(6000);
- send_stream->OnBitrateUpdated(update);
+ helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); });
}
TEST(AudioSendStreamTest, SSBweTargetInRangeRespected) {
@@ -559,7 +565,7 @@
Eq(DataRate::bps(helper.config().max_bitrate_bps - 5000)))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::bps(helper.config().max_bitrate_bps - 5000);
- send_stream->OnBitrateUpdated(update);
+ helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); });
}
TEST(AudioSendStreamTest, SSBweFieldTrialMinRespected) {
@@ -574,7 +580,7 @@
Eq(DataRate::kbps(6)))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::kbps(1);
- send_stream->OnBitrateUpdated(update);
+ helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); });
}
TEST(AudioSendStreamTest, SSBweFieldTrialMaxRespected) {
@@ -589,55 +595,64 @@
Eq(DataRate::kbps(64)))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::kbps(128);
- send_stream->OnBitrateUpdated(update);
+ helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); });
}
TEST(AudioSendStreamTest, SSBweWithOverhead) {
ScopedFieldTrials field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
- "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+ "WebRTC-SendSideBwe-WithOverhead/Enabled/"
+ "WebRTC-Audio-LegacyOverhead/Disabled/");
ConfigHelper helper(true, true);
auto send_stream = helper.CreateAudioSendStream();
+ EXPECT_CALL(*helper.channel_send(), CallEncoder(_)).Times(1);
+ send_stream->OnOverheadChanged(kOverheadPerPacket.bytes<size_t>());
const DataRate bitrate =
- DataRate::bps(helper.config().max_bitrate_bps) + kOverheadRate;
+ DataRate::bps(helper.config().max_bitrate_bps) + kMaxOverheadRate;
EXPECT_CALL(*helper.channel_send(),
OnBitrateAllocation(Field(
&BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
BitrateAllocationUpdate update;
update.target_bitrate = bitrate;
- send_stream->OnBitrateUpdated(update);
+ helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); });
}
TEST(AudioSendStreamTest, SSBweWithOverheadMinRespected) {
ScopedFieldTrials field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
"WebRTC-SendSideBwe-WithOverhead/Enabled/"
+ "WebRTC-Audio-LegacyOverhead/Disabled/"
"WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
ConfigHelper helper(true, true);
auto send_stream = helper.CreateAudioSendStream();
- const DataRate bitrate = DataRate::kbps(6) + kOverheadRate;
+ EXPECT_CALL(*helper.channel_send(), CallEncoder(_)).Times(1);
+ send_stream->OnOverheadChanged(kOverheadPerPacket.bytes<size_t>());
+ const DataRate bitrate = DataRate::kbps(6) + kMinOverheadRate;
EXPECT_CALL(*helper.channel_send(),
OnBitrateAllocation(Field(
&BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::kbps(1);
- send_stream->OnBitrateUpdated(update);
+ helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); });
}
TEST(AudioSendStreamTest, SSBweWithOverheadMaxRespected) {
ScopedFieldTrials field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
"WebRTC-SendSideBwe-WithOverhead/Enabled/"
+ "WebRTC-Audio-LegacyOverhead/Disabled/"
"WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
ConfigHelper helper(true, true);
auto send_stream = helper.CreateAudioSendStream();
- const DataRate bitrate = DataRate::kbps(64) + kOverheadRate;
+ EXPECT_CALL(*helper.channel_send(), CallEncoder(_)).Times(1);
+ send_stream->OnOverheadChanged(kOverheadPerPacket.bytes<size_t>());
+ const DataRate bitrate = DataRate::kbps(64) + kMaxOverheadRate;
EXPECT_CALL(*helper.channel_send(),
OnBitrateAllocation(Field(
&BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
BitrateAllocationUpdate update;
update.target_bitrate = DataRate::kbps(128);
- send_stream->OnBitrateUpdated(update);
+ helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); });
}
TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
@@ -652,7 +667,7 @@
update.packet_loss_ratio = 0;
update.round_trip_time = TimeDelta::ms(50);
update.bwe_period = TimeDelta::ms(5000);
- send_stream->OnBitrateUpdated(update);
+ helper.worker()->SendTask([&] { send_stream->OnBitrateUpdated(update); });
}
// Test that AudioSendStream doesn't recreate the encoder unnecessarily.