Revert of Adding BitrateController to audio network adaptor. (patchset #7 id:140001 of https://codereview.webrtc.org/2334613002/ )
Reason for revert:
ODR violation
Original issue's description:
> Adding BitrateController to audio network adaptor.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/26b039a137be0a8703766f45b546b29323de714f
> Cr-Commit-Position: refs/heads/master@{#14293}
TBR=michaelt@webrtc.org,henrik.lundin@webrtc.org,krasin@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2352223002
Cr-Commit-Position: refs/heads/master@{#14327}
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 09983c2..38aebcd 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -233,31 +233,6 @@
}
}
- rtc_source_set("audio_network_adaptor_unittests") {
- # Put sources for unittests of audio network adaptor in a separate
- # rtc_source_set to solve name collision on bitrate_controller_unittest.cc.
- testonly = true
- sources = [
- "audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
- "audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc",
- "audio_coding/audio_network_adaptor/channel_controller_unittest.cc",
- "audio_coding/audio_network_adaptor/controller_manager_unittest.cc",
- "audio_coding/audio_network_adaptor/dtx_controller_unittest.cc",
- "audio_coding/audio_network_adaptor/mock/mock_controller.h",
- "audio_coding/audio_network_adaptor/mock/mock_controller_manager.h",
- "audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc",
- ]
- deps = [
- "audio_coding:audio_network_adaptor",
- "//testing/gmock",
- "//testing/gtest",
- ]
- if (is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
- }
-
rtc_test("modules_unittests") {
testonly = true
@@ -272,6 +247,13 @@
"audio_coding/acm2/codec_manager_unittest.cc",
"audio_coding/acm2/initial_delay_manager_unittest.cc",
"audio_coding/acm2/rent_a_codec_unittest.cc",
+ "audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
+ "audio_coding/audio_network_adaptor/channel_controller_unittest.cc",
+ "audio_coding/audio_network_adaptor/controller_manager_unittest.cc",
+ "audio_coding/audio_network_adaptor/dtx_controller_unittest.cc",
+ "audio_coding/audio_network_adaptor/mock/mock_controller.h",
+ "audio_coding/audio_network_adaptor/mock/mock_controller_manager.h",
+ "audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc",
"audio_coding/codecs/audio_decoder_factory_unittest.cc",
"audio_coding/codecs/cng/audio_encoder_cng_unittest.cc",
"audio_coding/codecs/cng/cng_unittest.cc",
@@ -615,7 +597,6 @@
}
deps += [
- ":audio_network_adaptor_unittests",
"..:webrtc_common",
"../base:rtc_base", # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
"../common_audio",
@@ -629,6 +610,7 @@
"audio_coding",
"audio_coding:acm_receive_test",
"audio_coding:acm_send_test",
+ "audio_coding:audio_network_adaptor",
"audio_coding:builtin_audio_decoder_factory",
"audio_coding:cng",
"audio_coding:isac_fix",
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index a7b50a0..09202af 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -709,8 +709,6 @@
"audio_network_adaptor/audio_network_adaptor.cc",
"audio_network_adaptor/audio_network_adaptor_impl.cc",
"audio_network_adaptor/audio_network_adaptor_impl.h",
- "audio_network_adaptor/bitrate_controller.cc",
- "audio_network_adaptor/bitrate_controller.h",
"audio_network_adaptor/channel_controller.cc",
"audio_network_adaptor/channel_controller.h",
"audio_network_adaptor/controller.cc",
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi
index 2a591c3..d266ddb 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi
@@ -14,8 +14,6 @@
'audio_network_adaptor.cc',
'audio_network_adaptor_impl.cc',
'audio_network_adaptor_impl.h',
- 'bitrate_controller.h',
- 'bitrate_controller.cc',
'channel_controller.cc',
'channel_controller.h',
'controller.h',
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index 6562674..c17d389 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -28,7 +28,6 @@
MATCHER_P(NetworkMetricsIs, metric, "") {
return arg.uplink_bandwidth_bps == metric.uplink_bandwidth_bps &&
- arg.target_audio_bitrate_bps == metric.target_audio_bitrate_bps &&
arg.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction;
}
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
deleted file mode 100644
index 4cc49e8..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2016 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 "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
-
-#include <algorithm>
-
-#include "webrtc/base/checks.h"
-
-namespace webrtc {
-
-namespace {
-// TODO(minyue): consider passing this from a higher layer through
-// SetConstraints().
-// L2(14B) + IPv4(20B) + UDP(8B) + RTP(12B) + SRTP_AUTH(10B) = 64B = 512 bits
-constexpr int kPacketOverheadBits = 512;
-}
-
-BitrateController::Config::Config(int initial_bitrate_bps,
- int initial_frame_length_ms)
- : initial_bitrate_bps(initial_bitrate_bps),
- initial_frame_length_ms(initial_frame_length_ms) {}
-
-BitrateController::Config::~Config() = default;
-
-BitrateController::BitrateController(const Config& config)
- : config_(config),
- bitrate_bps_(config_.initial_bitrate_bps),
- overhead_rate_bps_(kPacketOverheadBits * 1000 /
- config_.initial_frame_length_ms) {
- RTC_DCHECK_GT(bitrate_bps_, 0);
- RTC_DCHECK_GT(overhead_rate_bps_, 0);
-}
-
-void BitrateController::MakeDecision(
- const NetworkMetrics& metrics,
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
- // Decision on |bitrate_bps| should not have been made.
- RTC_DCHECK(!config->bitrate_bps);
-
- if (metrics.target_audio_bitrate_bps) {
- int overhead_rate =
- config->frame_length_ms
- ? kPacketOverheadBits * 1000 / *config->frame_length_ms
- : overhead_rate_bps_;
- // If |metrics.target_audio_bitrate_bps| had included overhead, we would
- // simply do:
- // bitrate_bps_ = metrics.target_audio_bitrate_bps - overhead_rate;
- // Follow https://bugs.chromium.org/p/webrtc/issues/detail?id=6315 to track
- // progress regarding this.
- // Now we assume that |metrics.target_audio_bitrate_bps| can handle the
- // overhead of most recent packets.
- bitrate_bps_ = std::max(0, *metrics.target_audio_bitrate_bps +
- overhead_rate_bps_ - overhead_rate);
- // TODO(minyue): apply a smoothing on the |overhead_rate_bps_|.
- overhead_rate_bps_ = overhead_rate;
- }
- config->bitrate_bps = rtc::Optional<int>(bitrate_bps_);
-}
-
-} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
deleted file mode 100644
index cfc6fa8..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-
-namespace webrtc {
-
-class BitrateController final : public Controller {
- public:
- struct Config {
- Config(int initial_bitrate_bps, int frame_length_ms);
- ~Config();
- int initial_bitrate_bps;
- int initial_frame_length_ms;
- };
-
- explicit BitrateController(const Config& config);
-
- void MakeDecision(const NetworkMetrics& metrics,
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) override;
-
- private:
- const Config config_;
- int bitrate_bps_;
- int overhead_rate_bps_;
- RTC_DISALLOW_COPY_AND_ASSIGN(BitrateController);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
deleted file mode 100644
index 5a3a74a..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- * Copyright (c) 2016 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 "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
-
-namespace webrtc {
-
-namespace {
-
-// L2(14B) + IPv4(20B) + UDP(8B) + RTP(12B) + SRTP_AUTH(10B) = 64B = 512 bits
-constexpr int kPacketOverheadBits = 512;
-
-void CheckDecision(BitrateController* controller,
- const rtc::Optional<int>& target_audio_bitrate_bps,
- const rtc::Optional<int>& frame_length_ms,
- int expected_bitrate_bps) {
- Controller::NetworkMetrics metrics;
- metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
- config.frame_length_ms = frame_length_ms;
- controller->MakeDecision(metrics, &config);
- EXPECT_EQ(rtc::Optional<int>(expected_bitrate_bps), config.bitrate_bps);
-}
-
-} // namespace
-
-// These tests are named AnaBitrateControllerTest to distinguish from
-// BitrateControllerTest in
-// modules/bitrate_controller/bitrate_controller_unittest.cc.
-
-TEST(AnaBitrateControllerTest, OutputInitValueWhenTargetBitrateUnknown) {
- constexpr int kInitialBitrateBps = 32000;
- constexpr int kInitialFrameLengthMs = 20;
- BitrateController controller(
- BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
- CheckDecision(&controller, rtc::Optional<int>(),
- rtc::Optional<int>(kInitialFrameLengthMs * 2),
- kInitialBitrateBps);
-}
-
-TEST(AnaBitrateControllerTest, ChangeBitrateOnTargetBitrateChanged) {
- constexpr int kInitialBitrateBps = 32000;
- constexpr int kInitialFrameLengthMs = 20;
- BitrateController controller(
- BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
- constexpr int kTargetBitrateBps = 48000;
- // Frame length unchanged, bitrate changes in accordance with
- // |metrics.target_audio_bitrate_bps|
- CheckDecision(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<int>(kInitialFrameLengthMs), kTargetBitrateBps);
-}
-
-TEST(AnaBitrateControllerTest, TreatUnknownFrameLengthAsFrameLengthUnchanged) {
- constexpr int kInitialBitrateBps = 32000;
- constexpr int kInitialFrameLengthMs = 20;
- BitrateController controller(
- BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
- constexpr int kTargetBitrateBps = 48000;
- CheckDecision(&controller, rtc::Optional<int>(kTargetBitrateBps),
- rtc::Optional<int>(), kTargetBitrateBps);
-}
-
-TEST(AnaBitrateControllerTest, IncreaseBitrateOnFrameLengthIncreased) {
- constexpr int kInitialBitrateBps = 32000;
- constexpr int kInitialFrameLengthMs = 20;
- BitrateController controller(
- BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
- constexpr int kFrameLengthMs = 60;
- constexpr int kPacketOverheadRateDiff =
- kPacketOverheadBits * 1000 / kInitialFrameLengthMs -
- kPacketOverheadBits * 1000 / kFrameLengthMs;
- CheckDecision(&controller, rtc::Optional<int>(kInitialBitrateBps),
- rtc::Optional<int>(kFrameLengthMs),
- kInitialBitrateBps + kPacketOverheadRateDiff);
-}
-
-TEST(AnaBitrateControllerTest, DecreaseBitrateOnFrameLengthDecreased) {
- constexpr int kInitialBitrateBps = 32000;
- constexpr int kInitialFrameLengthMs = 60;
- BitrateController controller(
- BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
- constexpr int kFrameLengthMs = 20;
- constexpr int kPacketOverheadRateDiff =
- kPacketOverheadBits * 1000 / kInitialFrameLengthMs -
- kPacketOverheadBits * 1000 / kFrameLengthMs;
- CheckDecision(&controller, rtc::Optional<int>(kInitialBitrateBps),
- rtc::Optional<int>(kFrameLengthMs),
- kInitialBitrateBps + kPacketOverheadRateDiff);
-}
-
-TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
- constexpr int kInitialBitrateBps = 32000;
- constexpr int kInitialFrameLengthMs = 20;
- BitrateController controller(
- BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
-
- int last_overhead_bitrate =
- kPacketOverheadBits * 1000 / kInitialFrameLengthMs;
- int current_overhead_bitrate = kPacketOverheadBits * 1000 / 20;
- // Start from an arbitrary overall bitrate.
- int overall_bitrate = 34567;
- CheckDecision(
- &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
- rtc::Optional<int>(20), overall_bitrate - current_overhead_bitrate);
-
- // Next: increase overall bitrate.
- overall_bitrate += 100;
- CheckDecision(
- &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
- rtc::Optional<int>(20), overall_bitrate - current_overhead_bitrate);
-
- // Next: change frame length.
- current_overhead_bitrate = kPacketOverheadBits * 1000 / 60;
- CheckDecision(
- &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
- rtc::Optional<int>(60), overall_bitrate - current_overhead_bitrate);
- last_overhead_bitrate = current_overhead_bitrate;
-
- // Next: change frame length.
- current_overhead_bitrate = kPacketOverheadBits * 1000 / 20;
- CheckDecision(
- &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
- rtc::Optional<int>(20), overall_bitrate - current_overhead_bitrate);
- last_overhead_bitrate = current_overhead_bitrate;
-
- // Next: decrease overall bitrate and frame length.
- overall_bitrate -= 100;
- current_overhead_bitrate = kPacketOverheadBits * 1000 / 60;
- CheckDecision(
- &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
- rtc::Optional<int>(60), overall_bitrate - current_overhead_bitrate);
-}
-
-} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
index c1b16c7..f6a6079 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
@@ -23,7 +23,6 @@
~NetworkMetrics();
rtc::Optional<int> uplink_bandwidth_bps;
rtc::Optional<float> uplink_packet_loss_fraction;
- rtc::Optional<int> target_audio_bitrate_bps;
};
struct Constraints {
