Make ANA frame length controller more robust to encoder frame lengths.
Bug: webrtc:10820
Change-Id: Ic3a30976d0181de9cdd35e44d4c5439cadad4812
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149163
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28873}
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
index 3cb91fd..36e9eb9 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
@@ -75,10 +75,8 @@
RTC_DCHECK(!config->frame_length_ms);
if (FrameLengthIncreasingDecision(*config)) {
- ++frame_length_ms_;
prev_decision_increase_ = true;
} else if (FrameLengthDecreasingDecision(*config)) {
- --frame_length_ms_;
prev_decision_increase_ = false;
}
config->last_fl_change_increase = prev_decision_increase_;
@@ -99,7 +97,7 @@
}
bool FrameLengthController::FrameLengthIncreasingDecision(
- const AudioEncoderRuntimeConfig& config) const {
+ const AudioEncoderRuntimeConfig& config) {
// Increase frame length if
// 1. |uplink_bandwidth_bps| is known to be smaller or equal than
// |min_encoder_bitrate_bps| plus |prevent_overuse_margin_bps| plus the
@@ -108,12 +106,17 @@
// 3. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
// 4. |uplink_packet_loss_fraction| is known to be smaller than a threshold.
+ // Find next frame length to which a criterion is defined to shift from
+ // current frame length.
auto longer_frame_length_ms = std::next(frame_length_ms_);
- if (longer_frame_length_ms == config_.encoder_frame_lengths_ms.end())
- return false;
-
- auto increase_threshold = config_.fl_changing_bandwidths_bps.find(
- Config::FrameLengthChange(*frame_length_ms_, *longer_frame_length_ms));
+ auto increase_threshold = config_.fl_changing_bandwidths_bps.end();
+ while (longer_frame_length_ms != config_.encoder_frame_lengths_ms.end()) {
+ increase_threshold = config_.fl_changing_bandwidths_bps.find(
+ Config::FrameLengthChange(*frame_length_ms_, *longer_frame_length_ms));
+ if (increase_threshold != config_.fl_changing_bandwidths_bps.end())
+ break;
+ longer_frame_length_ms = std::next(longer_frame_length_ms);
+ }
if (increase_threshold == config_.fl_changing_bandwidths_bps.end())
return false;
@@ -134,18 +137,23 @@
OverheadRateBps(*overhead_bytes_per_packet_ +
config_.fl_increase_overhead_offset,
*frame_length_ms_)) {
+ frame_length_ms_ = longer_frame_length_ms;
return true;
}
- return (uplink_bandwidth_bps_ &&
- *uplink_bandwidth_bps_ <= increase_threshold->second) &&
- (uplink_packet_loss_fraction_ &&
- *uplink_packet_loss_fraction_ <=
- config_.fl_increasing_packet_loss_fraction);
+ if ((uplink_bandwidth_bps_ &&
+ *uplink_bandwidth_bps_ <= increase_threshold->second) &&
+ (uplink_packet_loss_fraction_ &&
+ *uplink_packet_loss_fraction_ <=
+ config_.fl_increasing_packet_loss_fraction)) {
+ frame_length_ms_ = longer_frame_length_ms;
+ return true;
+ }
+ return false;
}
bool FrameLengthController::FrameLengthDecreasingDecision(
- const AudioEncoderRuntimeConfig& config) const {
+ const AudioEncoderRuntimeConfig& config) {
// Decrease frame length if
// 1. shorter frame length is available AND
// 2. |uplink_bandwidth_bps| is known to be bigger than
@@ -154,12 +162,18 @@
// one or more of the followings:
// 3. |uplink_bandwidth_bps| is known to be larger than a threshold,
// 4. |uplink_packet_loss_fraction| is known to be larger than a threshold,
- if (frame_length_ms_ == config_.encoder_frame_lengths_ms.begin())
- return false;
- auto shorter_frame_length_ms = std::prev(frame_length_ms_);
- auto decrease_threshold = config_.fl_changing_bandwidths_bps.find(
- Config::FrameLengthChange(*frame_length_ms_, *shorter_frame_length_ms));
+ // Find next frame length to which a criterion is defined to shift from
+ // current frame length.
+ auto shorter_frame_length_ms = frame_length_ms_;
+ auto decrease_threshold = config_.fl_changing_bandwidths_bps.end();
+ while (shorter_frame_length_ms != config_.encoder_frame_lengths_ms.begin()) {
+ shorter_frame_length_ms = std::prev(shorter_frame_length_ms);
+ decrease_threshold = config_.fl_changing_bandwidths_bps.find(
+ Config::FrameLengthChange(*frame_length_ms_, *shorter_frame_length_ms));
+ if (decrease_threshold != config_.fl_changing_bandwidths_bps.end())
+ break;
+ }
if (decrease_threshold == config_.fl_changing_bandwidths_bps.end())
return false;
@@ -173,11 +187,15 @@
return false;
}
- return (uplink_bandwidth_bps_ &&
- *uplink_bandwidth_bps_ >= decrease_threshold->second) ||
- (uplink_packet_loss_fraction_ &&
- *uplink_packet_loss_fraction_ >=
- config_.fl_decreasing_packet_loss_fraction);
+ if ((uplink_bandwidth_bps_ &&
+ *uplink_bandwidth_bps_ >= decrease_threshold->second) ||
+ (uplink_packet_loss_fraction_ &&
+ *uplink_packet_loss_fraction_ >=
+ config_.fl_decreasing_packet_loss_fraction)) {
+ frame_length_ms_ = shorter_frame_length_ms;
+ return true;
+ }
+ return false;
}
} // namespace webrtc
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
index 0268ddc..74a787e 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
@@ -67,11 +67,9 @@
void MakeDecision(AudioEncoderRuntimeConfig* config) override;
private:
- bool FrameLengthIncreasingDecision(
- const AudioEncoderRuntimeConfig& config) const;
+ bool FrameLengthIncreasingDecision(const AudioEncoderRuntimeConfig& config);
- bool FrameLengthDecreasingDecision(
- const AudioEncoderRuntimeConfig& config) const;
+ bool FrameLengthDecreasingDecision(const AudioEncoderRuntimeConfig& config);
const Config config_;
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
index 6709336..9db9853 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
@@ -34,6 +34,7 @@
(kFl60msTo20msBandwidthBps + kFl20msTo60msBandwidthBps) / 2;
constexpr float kMediumPacketLossFraction =
(kFlDecreasingPacketLossFraction + kFlIncreasingPacketLossFraction) / 2;
+const std::set<int> kDefaultEncoderFrameLengthsMs = {20, 40, 60, 120};
int VeryLowBitrate(int frame_length_ms) {
return kMinEncoderBitrateBps + kPreventOveruseMarginBps +
@@ -112,16 +113,16 @@
} // namespace
TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkBandwidth) {
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 60);
UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
absl::nullopt, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 20);
}
TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkPacketLossFraction) {
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 60);
UpdateNetworkMetrics(controller.get(), absl::nullopt,
kFlDecreasingPacketLossFraction,
kOverheadBytesPerPacket);
@@ -142,8 +143,8 @@
// 1. |uplink_bandwidth_bps| is at medium level,
// 2. |uplink_packet_loss_fraction| is at medium,
// 3. FEC is not decided ON.
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 60);
UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
kMediumPacketLossFraction, kOverheadBytesPerPacket);
CheckDecision(controller.get(), 60);
@@ -155,8 +156,8 @@
// 2. |uplink_packet_loss_fraction| is known to be smaller than a threshold
// AND
// 3. FEC is not decided or OFF.
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 20);
UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
@@ -164,8 +165,8 @@
}
TEST(FrameLengthControllerTest, IncreaseTo60MsOnVeryLowUplinkBandwidth) {
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 20);
// We set packet loss fraction to kFlDecreasingPacketLossFraction, which
// should have prevented frame length to increase, if the uplink bandwidth
// was not this low.
@@ -176,8 +177,8 @@
}
TEST(FrameLengthControllerTest, Maintain60MsOnVeryLowUplinkBandwidth) {
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 60);
// We set packet loss fraction to FlDecreasingPacketLossFraction, which should
// have caused the frame length to decrease, if the uplink bandwidth was not
// this low.
@@ -195,8 +196,8 @@
// FrameLengthController::UpdateNetworkMetrics(...) can handle multiple
// network updates at once. This is, however, not a common use case in current
// audio_network_adaptor_impl.cc.
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 20);
Controller::NetworkMetrics network_metrics;
network_metrics.uplink_bandwidth_bps = kFl20msTo60msBandwidthBps;
network_metrics.uplink_packet_loss_fraction = kFlIncreasingPacketLossFraction;
@@ -217,8 +218,8 @@
}
TEST(FrameLengthControllerTest, Maintain20MsOnMediumUplinkBandwidth) {
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 20);
UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
@@ -226,8 +227,8 @@
}
TEST(FrameLengthControllerTest, Maintain20MsOnMediumUplinkPacketLossFraction) {
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 20);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 20);
// Use a low uplink bandwidth that would cause frame length to increase if
// uplink packet loss fraction was low.
UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
@@ -236,8 +237,8 @@
}
TEST(FrameLengthControllerTest, Maintain60MsWhenNo120msCriteriaIsSet) {
- auto controller =
- CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60, 120}, 60);
+ auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+ kDefaultEncoderFrameLengthsMs, 60);
UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
@@ -246,7 +247,7 @@
TEST(FrameLengthControllerTest, From120MsTo20MsOnHighUplinkBandwidth) {
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
- {20, 60, 120}, 120);
+ kDefaultEncoderFrameLengthsMs, 120);
// It takes two steps for frame length to go from 120ms to 20ms.
UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
absl::nullopt, kOverheadBytesPerPacket);
@@ -259,7 +260,7 @@
TEST(FrameLengthControllerTest, From120MsTo20MsOnHighUplinkPacketLossFraction) {
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
- {20, 60, 120}, 120);
+ kDefaultEncoderFrameLengthsMs, 120);
// It takes two steps for frame length to go from 120ms to 20ms.
UpdateNetworkMetrics(controller.get(), absl::nullopt,
kFlDecreasingPacketLossFraction,
@@ -274,7 +275,7 @@
TEST(FrameLengthControllerTest, Maintain120MsOnVeryLowUplinkBandwidth) {
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
- {20, 60, 120}, 120);
+ kDefaultEncoderFrameLengthsMs, 120);
// We set packet loss fraction to FlDecreasingPacketLossFraction, which should
// have caused the frame length to decrease, if the uplink bandwidth was not
// this low.
@@ -286,7 +287,7 @@
TEST(FrameLengthControllerTest, From60MsTo120MsOnVeryLowUplinkBandwidth) {
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
- {20, 60, 120}, 60);
+ kDefaultEncoderFrameLengthsMs, 60);
// We set packet loss fraction to FlDecreasingPacketLossFraction, which should
// have prevented frame length to increase, if the uplink bandwidth was not
// this low.
@@ -301,7 +302,7 @@
// 1. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
// 2. |uplink_packet_loss_fraction| is known to be smaller than a threshold.
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
- {20, 60, 120}, 20);
+ kDefaultEncoderFrameLengthsMs, 20);
// It takes two steps for frame length to go from 20ms to 120ms.
UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
kFlIncreasingPacketLossFraction,
@@ -328,7 +329,7 @@
TEST(FrameLengthControllerTest, CheckBehaviorOnChangingNetworkMetrics) {
auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
- {20, 60, 120}, 20);
+ kDefaultEncoderFrameLengthsMs, 20);
UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
kFlIncreasingPacketLossFraction,
kOverheadBytesPerPacket);
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index d15a242..f901d3c 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -55,10 +55,10 @@
// These two lists must be sorted from low to high
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-constexpr int kANASupportedFrameLengths[] = {20, 60, 120};
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60, 120};
constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120};
#else
-constexpr int kANASupportedFrameLengths[] = {20, 60};
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60};
constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
#endif
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 8ae9ee7..3870ecd 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -334,9 +334,11 @@
ElementsAre(states->encoder->next_frame_length_ms()));
states->encoder->SetReceiverFrameLengthRange(0, 12345);
states->encoder->SetReceiverFrameLengthRange(21, 60);
- EXPECT_THAT(states->encoder->supported_frame_lengths_ms(), ElementsAre(60));
+ EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+ ElementsAre(40, 60));
states->encoder->SetReceiverFrameLengthRange(20, 59);
- EXPECT_THAT(states->encoder->supported_frame_lengths_ms(), ElementsAre(20));
+ EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+ ElementsAre(20, 40));
}
TEST_P(AudioEncoderOpusTest,
@@ -780,9 +782,9 @@
const webrtc::SdpAudioFormat format("opus", 48000, 2);
const auto default_config = *AudioEncoderOpus::SdpToConfig(format);
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
- const std::vector<int> default_supported_frame_lengths_ms({20, 60, 120});
+ const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60, 120});
#else
- const std::vector<int> default_supported_frame_lengths_ms({20, 60});
+ const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60});
#endif
AudioEncoderOpusConfig config;
