Adds WebRTC-Audio-PriorityBitrate for controlling audio/video rate allocation

Bug: webrtc:15769
Change-Id: Id260fb9540ecb79b0c349937e55db343e04178c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334702
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Cr-Commit-Position: refs/heads/main@{#41631}

audio/audio_send_stream.cc

@@ -157,6 +157,8 @@ AudioSendStream::AudioSendStream(
       event_log_(event_log),
       use_legacy_overhead_calculation_(
           field_trials_.IsEnabled("WebRTC-Audio-LegacyOverhead")),
+      enable_priority_bitrate_(
+          !field_trials_.IsDisabled("WebRTC-Audio-PriorityBitrate")),
       bitrate_allocator_(bitrate_allocator),
       rtp_transport_(rtp_transport),
       rtp_rtcp_module_(channel_send_->GetRtpRtcp()),
@@ -807,6 +809,10 @@ void AudioSendStream::ConfigureBitrateObserver() {
     priority_bitrate = *allocation_settings_.priority_bitrate_raw;
   }
 
+  if (!enable_priority_bitrate_) {
+    priority_bitrate = DataRate::BitsPerSec(0);
+  }
+
   bitrate_allocator_->AddObserver(
       this,
       MediaStreamAllocationConfig{

audio/audio_send_stream.h

@@ -173,6 +173,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   const std::unique_ptr<voe::ChannelSendInterface> channel_send_;
   RtcEventLog* const event_log_;
   const bool use_legacy_overhead_calculation_;
+  const bool enable_priority_bitrate_;
 
   int encoder_sample_rate_hz_ RTC_GUARDED_BY(worker_thread_checker_) = 0;
   size_t encoder_num_channels_ RTC_GUARDED_BY(worker_thread_checker_) = 0;

audio/audio_send_stream_unittest.cc

@@ -21,6 +21,7 @@
 #include "audio/audio_state.h"
 #include "audio/conversion.h"
 #include "audio/mock_voe_channel_proxy.h"
+#include "call/test/mock_bitrate_allocator.h"
 #include "call/test/mock_rtp_transport_controller_send.h"
 #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "modules/audio_device/include/mock_audio_device.h"
@@ -155,7 +156,6 @@ struct ConfigHelper {
             use_null_audio_processing
                 ? nullptr
                 : rtc::make_ref_counted<NiceMock<MockAudioProcessing>>()),
-        bitrate_allocator_(&limit_observer_),
         audio_encoder_(nullptr) {
     using ::testing::Invoke;
 
@@ -203,6 +203,7 @@ struct ConfigHelper {
   MockRtpRtcpInterface* rtp_rtcp() { return &rtp_rtcp_; }
   MockChannelSend* channel_send() { return channel_send_; }
   RtpTransportControllerSendInterface* transport() { return &rtp_transport_; }
+  MockBitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
 
   static void AddBweToConfig(AudioSendStream::Config* config) {
     config->rtp.extensions.push_back(RtpExtension(
@@ -328,7 +329,7 @@ struct ConfigHelper {
   ::testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_;
   ::testing::NiceMock<MockRtpRtcpInterface> rtp_rtcp_;
   ::testing::NiceMock<MockLimitObserver> limit_observer_;
-  BitrateAllocator bitrate_allocator_;
+  ::testing::NiceMock<MockBitrateAllocator> bitrate_allocator_;
   std::unique_ptr<AudioEncoder> audio_encoder_;
 };
 
@@ -924,5 +925,39 @@ TEST(AudioSendStreamTest, ReconfigureWithFrameEncryptor) {
     send_stream->Reconfigure(new_config, nullptr);
   }
 }
+
+TEST(AudioSendStreamTest, DefaultsHonorsPriorityBitrate) {
+  ConfigHelper helper(true, true, true);
+  ScopedKeyValueConfig field_trials(helper.field_trials,
+                                    "WebRTC-Audio-Allocation/prio_rate:20/");
+  auto send_stream = helper.CreateAudioSendStream();
+  EXPECT_CALL(*helper.bitrate_allocator(), AddObserver(send_stream.get(), _))
+      .WillOnce(Invoke(
+          [&](BitrateAllocatorObserver*, MediaStreamAllocationConfig config) {
+            EXPECT_EQ(config.priority_bitrate_bps, 20000);
+          }));
+  EXPECT_CALL(*helper.channel_send(), StartSend());
+  send_stream->Start();
+  EXPECT_CALL(*helper.channel_send(), StopSend());
+  send_stream->Stop();
+}
+
+TEST(AudioSendStreamTest, OverridesPriorityBitrate) {
+  ConfigHelper helper(true, true, true);
+  ScopedKeyValueConfig field_trials(helper.field_trials,
+                                    "WebRTC-Audio-Allocation/prio_rate:20/"
+                                    "WebRTC-Audio-PriorityBitrate/Disabled/");
+  auto send_stream = helper.CreateAudioSendStream();
+  EXPECT_CALL(*helper.bitrate_allocator(), AddObserver(send_stream.get(), _))
+      .WillOnce(Invoke(
+          [&](BitrateAllocatorObserver*, MediaStreamAllocationConfig config) {
+            EXPECT_EQ(config.priority_bitrate_bps, 0);
+          }));
+  EXPECT_CALL(*helper.channel_send(), StartSend());
+  send_stream->Start();
+  EXPECT_CALL(*helper.channel_send(), StopSend());
+  send_stream->Stop();
+}
+
 }  // namespace test
 }  // namespace webrtc

experiments/field_trials.py

@@ -53,6 +53,9 @@ ACTIVE_FIELD_TRIALS: FrozenSet[FieldTrial] = frozenset([
     FieldTrial('WebRTC-Audio-OpusGeneratePlc',
                'webrtc:13322',
                date(2024, 4, 1)),
+    FieldTrial('WebRTC-Audio-PriorityBitrate',
+               'webrtc:15769',
+               date(2024, 4, 1)),
     FieldTrial('WebRTC-AV1-OverridePriorityBitrate',
                'webrtc:15763',
                date(2024, 4, 1)),