Add a simple frame length controller.
This will be used when adaptivePtime is enabled.
Bug: chromium:1086942
Change-Id: I63c947c53a8c5b8e0825b78b847c3f7900197d6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177421
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31544}
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 9902361..eee17d5 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -882,6 +882,8 @@
"audio_network_adaptor/fec_controller_plr_based.h",
"audio_network_adaptor/frame_length_controller.cc",
"audio_network_adaptor/frame_length_controller.h",
+ "audio_network_adaptor/frame_length_controller_v2.cc",
+ "audio_network_adaptor/frame_length_controller_v2.h",
"audio_network_adaptor/include/audio_network_adaptor.h",
"audio_network_adaptor/util/threshold_curve.h",
]
@@ -902,7 +904,10 @@
"../../system_wrappers",
"../../system_wrappers:field_trial",
]
- absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/algorithm:container",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
if (rtc_enable_protobuf) {
deps += [
@@ -1931,6 +1936,7 @@
"audio_network_adaptor/event_log_writer_unittest.cc",
"audio_network_adaptor/fec_controller_plr_based_unittest.cc",
"audio_network_adaptor/frame_length_controller_unittest.cc",
+ "audio_network_adaptor/frame_length_controller_v2_unittest.cc",
"audio_network_adaptor/util/threshold_curve_unittest.cc",
"codecs/builtin_audio_decoder_factory_unittest.cc",
"codecs/builtin_audio_encoder_factory_unittest.cc",
diff --git a/modules/audio_coding/audio_network_adaptor/config.proto b/modules/audio_coding/audio_network_adaptor/config.proto
index 90c58e5..347372e 100644
--- a/modules/audio_coding/audio_network_adaptor/config.proto
+++ b/modules/audio_coding/audio_network_adaptor/config.proto
@@ -1,8 +1,10 @@
syntax = "proto2";
+
+package webrtc.audio_network_adaptor.config;
+
option optimize_for = LITE_RUNTIME;
option java_package = "org.webrtc.AudioNetworkAdaptor";
option java_outer_classname = "Config";
-package webrtc.audio_network_adaptor.config;
message FecController {
message Threshold {
@@ -116,6 +118,19 @@
optional int32 fl_60ms_to_40ms_bandwidth_bps = 12;
}
+message FrameLengthControllerV2 {
+ // FrameLengthControllerV2 chooses the frame length by taking the target
+ // bitrate and subtracting the overhead bitrate to obtain the remaining
+ // bitrate for the payload. The chosen frame length is the shortest possible
+ // where the payload bitrate is more than |min_payload_bitrate_bps|.
+ optional int32 min_payload_bitrate_bps = 1;
+
+ // If true, uses the stable target bitrate to decide the frame length. This
+ // will result in less frame length toggling but spending more time at longer
+ // frame lengths compared to using the normal target bitrate.
+ optional bool use_slow_adaptation = 2;
+}
+
message ChannelController {
// Uplink bandwidth above which the number of encoded channels should switch
// from 1 to 2.
@@ -164,6 +179,7 @@
DtxController dtx_controller = 24;
BitrateController bitrate_controller = 25;
FecControllerRplrBased fec_controller_rplr_based = 26;
+ FrameLengthControllerV2 frame_length_controller_v2 = 27;
}
}
@@ -177,4 +193,3 @@
// made.
optional float min_reordering_squared_distance = 3;
}
-
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index c7aad1d..415b9fc 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -11,6 +11,7 @@
#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include <cmath>
+#include <memory>
#include <string>
#include <utility>
@@ -20,6 +21,7 @@
#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h"
#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/logging.h"
@@ -197,6 +199,14 @@
initial_bitrate_bps, initial_frame_length_ms,
fl_increase_overhead_offset, fl_decrease_overhead_offset)));
}
+
+std::unique_ptr<FrameLengthControllerV2> CreateFrameLengthControllerV2(
+ const audio_network_adaptor::config::FrameLengthControllerV2& config,
+ rtc::ArrayView<const int> encoder_frame_lengths_ms) {
+ return std::make_unique<FrameLengthControllerV2>(
+ encoder_frame_lengths_ms, config.min_payload_bitrate_bps(),
+ config.use_slow_adaptation());
+}
#endif // WEBRTC_ENABLE_PROTOBUF
} // namespace
@@ -277,6 +287,11 @@
controller_config.bitrate_controller(), initial_bitrate_bps,
initial_frame_length_ms);
break;
+ case audio_network_adaptor::config::Controller::kFrameLengthControllerV2:
+ controller = CreateFrameLengthControllerV2(
+ controller_config.frame_length_controller_v2(),
+ encoder_frame_lengths_ms);
+ break;
default:
RTC_NOTREACHED();
}
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index 4286434..c71bbc9 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -260,6 +260,14 @@
kChracteristicPacketLossFraction[1]);
}
+void AddFrameLengthControllerV2Config(
+ audio_network_adaptor::config::ControllerManager* config) {
+ auto controller =
+ config->add_controllers()->mutable_frame_length_controller_v2();
+ controller->set_min_payload_bitrate_bps(16000);
+ controller->set_use_slow_adaptation(true);
+}
+
constexpr int kInitialBitrateBps = 24000;
constexpr size_t kIntialChannelsToEncode = 1;
constexpr bool kInitialDtxEnabled = true;
@@ -464,6 +472,14 @@
ControllerType::CHANNEL, ControllerType::DTX,
ControllerType::BIT_RATE});
}
+
+TEST(ControllerManagerTest, CreateFrameLengthControllerV2) {
+ audio_network_adaptor::config::ControllerManager config;
+ AddFrameLengthControllerV2Config(&config);
+ auto states = CreateControllerManager(config.SerializeAsString());
+ auto controllers = states.controller_manager->GetControllers();
+ EXPECT_TRUE(controllers.size() == 1);
+}
#endif // WEBRTC_ENABLE_PROTOBUF
} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc
new file mode 100644
index 0000000..36fc10b
--- /dev/null
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc
@@ -0,0 +1,73 @@
+/*
+ * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h"
+
+#include <algorithm>
+
+#include "absl/algorithm/container.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+int OverheadBps(int overhead_bytes_per_packet, int frame_length_ms) {
+ return overhead_bytes_per_packet * 8 * 1000 / frame_length_ms;
+}
+
+} // namespace
+
+FrameLengthControllerV2::FrameLengthControllerV2(
+ rtc::ArrayView<const int> encoder_frame_lengths_ms,
+ int min_payload_bitrate_bps,
+ bool use_slow_adaptation)
+ : encoder_frame_lengths_ms_(encoder_frame_lengths_ms.begin(),
+ encoder_frame_lengths_ms.end()),
+ min_payload_bitrate_bps_(min_payload_bitrate_bps),
+ use_slow_adaptation_(use_slow_adaptation) {
+ RTC_CHECK(!encoder_frame_lengths_ms_.empty());
+ absl::c_sort(encoder_frame_lengths_ms_);
+}
+
+void FrameLengthControllerV2::UpdateNetworkMetrics(
+ const NetworkMetrics& network_metrics) {
+ if (network_metrics.target_audio_bitrate_bps) {
+ target_bitrate_bps_ = network_metrics.target_audio_bitrate_bps;
+ }
+ if (network_metrics.overhead_bytes_per_packet) {
+ overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet;
+ }
+ if (network_metrics.uplink_bandwidth_bps) {
+ uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+ }
+}
+
+void FrameLengthControllerV2::MakeDecision(AudioEncoderRuntimeConfig* config) {
+ if (!target_bitrate_bps_ || !overhead_bytes_per_packet_ ||
+ !uplink_bandwidth_bps_) {
+ return;
+ }
+
+ auto it =
+ absl::c_find_if(encoder_frame_lengths_ms_, [&](int frame_length_ms) {
+ int target = use_slow_adaptation_ ? *uplink_bandwidth_bps_
+ : *target_bitrate_bps_;
+ return target -
+ OverheadBps(*overhead_bytes_per_packet_, frame_length_ms) >
+ min_payload_bitrate_bps_;
+ });
+
+ // Longest frame length is chosen if none match our criteria.
+ config->frame_length_ms = it != encoder_frame_lengths_ms_.end()
+ ? *it
+ : encoder_frame_lengths_ms_.back();
+}
+
+} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
new file mode 100644
index 0000000..d7102b0
--- /dev/null
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_V2_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_V2_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+
+namespace webrtc {
+
+class FrameLengthControllerV2 final : public Controller {
+ public:
+ FrameLengthControllerV2(rtc::ArrayView<const int> encoder_frame_lengths_ms,
+ int min_payload_bitrate_bps,
+ bool use_slow_adaptation);
+
+ void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+ void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+ std::vector<int> encoder_frame_lengths_ms_;
+ const int min_payload_bitrate_bps_;
+ const bool use_slow_adaptation_;
+
+ absl::optional<int> uplink_bandwidth_bps_;
+ absl::optional<int> target_bitrate_bps_;
+ absl::optional<int> overhead_bytes_per_packet_;
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_V2_H_
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc
new file mode 100644
index 0000000..1c88f47
--- /dev/null
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc
@@ -0,0 +1,121 @@
+/*
+ * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60, 120};
+constexpr int kMinPayloadBitrateBps = 16000;
+
+} // namespace
+
+class FrameLengthControllerV2Test : public testing::Test {
+ protected:
+ AudioEncoderRuntimeConfig GetDecision() {
+ AudioEncoderRuntimeConfig config;
+ controller_->MakeDecision(&config);
+ return config;
+ }
+
+ void SetOverhead(int overhead_bytes_per_packet) {
+ overhead_bytes_per_packet_ = overhead_bytes_per_packet;
+ Controller::NetworkMetrics metrics;
+ metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
+ controller_->UpdateNetworkMetrics(metrics);
+ }
+
+ void SetTargetBitrate(int target_audio_bitrate_bps) {
+ target_audio_bitrate_bps_ = target_audio_bitrate_bps;
+ Controller::NetworkMetrics metrics;
+ metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
+ controller_->UpdateNetworkMetrics(metrics);
+ }
+
+ void SetUplinkBandwidth(int uplink_bandwidth_bps) {
+ Controller::NetworkMetrics metrics;
+ metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+ controller_->UpdateNetworkMetrics(metrics);
+ }
+
+ void ExpectFrameLengthDecision(int expected_frame_length_ms) {
+ auto config = GetDecision();
+ EXPECT_EQ(*config.frame_length_ms, expected_frame_length_ms);
+ }
+
+ std::unique_ptr<FrameLengthControllerV2> controller_ =
+ std::make_unique<FrameLengthControllerV2>(kANASupportedFrameLengths,
+ kMinPayloadBitrateBps,
+ /*use_slow_adaptation=*/false);
+ absl::optional<int> target_audio_bitrate_bps_;
+ absl::optional<int> overhead_bytes_per_packet_;
+};
+
+// Don't return any decision if we haven't received all required network
+// metrics.
+TEST_F(FrameLengthControllerV2Test, RequireNetworkMetrics) {
+ auto config = GetDecision();
+ EXPECT_FALSE(config.bitrate_bps);
+ EXPECT_FALSE(config.frame_length_ms);
+
+ SetOverhead(30);
+ config = GetDecision();
+ EXPECT_FALSE(config.frame_length_ms);
+
+ SetTargetBitrate(32000);
+ config = GetDecision();
+ EXPECT_FALSE(config.frame_length_ms);
+
+ SetUplinkBandwidth(32000);
+ config = GetDecision();
+ EXPECT_TRUE(config.frame_length_ms);
+}
+
+TEST_F(FrameLengthControllerV2Test, UseFastAdaptation) {
+ SetOverhead(50);
+ SetTargetBitrate(50000);
+ SetUplinkBandwidth(50000);
+ ExpectFrameLengthDecision(20);
+
+ SetTargetBitrate(20000);
+ ExpectFrameLengthDecision(120);
+
+ SetTargetBitrate(30000);
+ ExpectFrameLengthDecision(40);
+
+ SetTargetBitrate(25000);
+ ExpectFrameLengthDecision(60);
+}
+
+TEST_F(FrameLengthControllerV2Test, UseSlowAdaptation) {
+ controller_ = std::make_unique<FrameLengthControllerV2>(
+ kANASupportedFrameLengths, kMinPayloadBitrateBps,
+ /*use_slow_adaptation=*/true);
+ SetOverhead(50);
+ SetTargetBitrate(50000);
+ SetUplinkBandwidth(20000);
+ ExpectFrameLengthDecision(120);
+
+ SetUplinkBandwidth(30000);
+ ExpectFrameLengthDecision(40);
+
+ SetUplinkBandwidth(40000);
+ ExpectFrameLengthDecision(20);
+}
+
+} // namespace webrtc
