Revert "Change buffer level filter to store current level in number of samples."
This reverts commit 87977dd06e702ed517f26235c12e37bd927527c7.
Reason for revert: Breaks downstream project
Original change's description:
> Change buffer level filter to store current level in number of samples.
>
> The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
>
> Bug: webrtc:10736
> Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28368}
TBR=henrik.lundin@webrtc.org,minyue@webrtc.org,jakobi@webrtc.org
Change-Id: I3900c9f6071fce51d13fb3b7c886157304d7a5c3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143786
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28369}
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index f9488a3..c95091f 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -994,35 +994,35 @@
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC)
TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
- Run(8000, PlatformChecksum("bc5cc2391174c865f02d9b2ae568020a",
- "06621315b131e5ece54aff186ec7b9e4",
- "2d05bd35f4dd9798e4189d7da6e6ba10",
+ Run(8000, PlatformChecksum("bcfbe2e89b4317b22e29557168edf187",
+ "af15addb648cf7f032d6415672365fb3",
+ "54a0008eb79537dee1d8fdaa5bc29f4b",
"4598140b5e4f7ee66c5adad609e65a3e",
- "c0c14f9d8d4151510cafdaf78e2ebd20"));
+ "3155d7f2593a3276986f36221a61783c"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
- Run(16000, PlatformChecksum("9e3893aeb5dda4f9cea6adec21d21ad8",
- "70d26c18385e05a35162460a2ef6a0e4",
- "b1ef73d7dde40033e30a76a1cc6a1ab6",
+ Run(16000, PlatformChecksum("1737deef193e6c90e139ce82b7361ae4",
+ "9e2a9f7728c71d6559ce3a32d2b10a5d",
+ "114958862099142ac78b12100c21cb8d",
"f2aad418af974a3b1694d5ae5cc2c3c7",
- "25ae85ea1d2d629956cec7ea0b94f51c"));
+ "af2889a5ca84fb40c9aa209b9318ee7a"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
- Run(32000, PlatformChecksum("4be3e35ebfdce55157491945610ef663",
- "424fcd896d2050530648de083567aff7",
- "ab50b8546f56c359c8e4c7d74e49aae4",
+ Run(32000, PlatformChecksum("1bf40ff024c6aa5b832d1d242c29cb3b",
+ "3c9690cd136e9ecd1b26a22f70fe1d5c",
+ "a1a3a01d8e25fcd11f1cedcd02e968b8",
"100869c8dcde51346c2073e52a272d98",
- "87731746b160e4ae8313fa8af92bba12"));
+ "33695077e9ec6bca80819ce2ba263a78"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
- Run(48000, PlatformChecksum("1df82e2f5c1c96d94ad0edc55c723d01",
- "a623e44f7e41263245361e2b8c6cedd2",
- "22d0e9c2b62f367fb2d2052569c70e55",
+ Run(48000, PlatformChecksum("bf92db1e502deff5adf6fd2e6ab9a2e5",
+ "c37b110ab50d87620972daee5d1eaf31",
+ "5d55b68be7bcf39b60fcc74519363fb4",
"bd44bf97e7899186532f91235cef444d",
- "c3188d9fdaa8057a98815c9b7ecc48f1"));
+ "32eec738698ffe62b9777d6a349cd596"));
}
TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
@@ -1105,11 +1105,11 @@
rtc::scoped_refptr<rtc::RefCountedObject<ADFactory>> factory(
new rtc::RefCountedObject<ADFactory>);
Run(48000,
- PlatformChecksum("1df82e2f5c1c96d94ad0edc55c723d01",
- "a623e44f7e41263245361e2b8c6cedd2",
- "22d0e9c2b62f367fb2d2052569c70e55",
+ PlatformChecksum("bf92db1e502deff5adf6fd2e6ab9a2e5",
+ "c37b110ab50d87620972daee5d1eaf31",
+ "5d55b68be7bcf39b60fcc74519363fb4",
"bd44bf97e7899186532f91235cef444d",
- "c3188d9fdaa8057a98815c9b7ecc48f1"),
+ "32eec738698ffe62b9777d6a349cd596"),
factory, [](AudioCodingModule* acm) {
acm->SetReceiveCodecs({{0, {"MockPCMu", 8000, 1}},
{103, {"ISAC", 16000, 1}},
@@ -1328,7 +1328,7 @@
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
"2c9cb15d4ed55b5a0cadd04883bc73b0",
"9336a9b993cbd8a751f0e8958e66c89c",
- "5c2eb46199994506236f68b2c8e51b0d",
+ "bd4682225f7c4ad5f2049f6769713ac2",
"343f1f42be0607c61e6516aece424609",
"2c9cb15d4ed55b5a0cadd04883bc73b0"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
@@ -1343,11 +1343,11 @@
TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) {
ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960));
Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
- "f59760fa000991ee5fa81f2e607db254",
- "986aa16d7097a26e32e212e39ec58517",
- "9a81e467eb1485f84aca796f8ea65011",
+ "1ad29139a04782a33daad8c2b9b35875",
+ "14d63c5f08127d280e722e3191b73bdd",
+ "edcf26694c289e3d9691faf79b74f09f",
"ef75e900e6f375e3061163c53fd09a63",
- "f59760fa000991ee5fa81f2e607db254"),
+ "1ad29139a04782a33daad8c2b9b35875"),
AcmReceiverBitExactnessOldApi::PlatformChecksum(
"9e0a0ab743ad987b55b8e14802769c56",
"ebe04a819d3a9d83a83a17f271e1139a",
diff --git a/modules/audio_coding/neteq/buffer_level_filter.cc b/modules/audio_coding/neteq/buffer_level_filter.cc
index 675091d..2f96618 100644
--- a/modules/audio_coding/neteq/buffer_level_filter.cc
+++ b/modules/audio_coding/neteq/buffer_level_filter.cc
@@ -22,31 +22,52 @@
}
void BufferLevelFilter::Reset() {
- filtered_current_level_ = 0.0;
- level_factor_ = 0.988;
+ filtered_current_level_ = 0;
+ level_factor_ = 253;
}
-void BufferLevelFilter::Update(size_t buffer_size_samples,
- int time_stretched_samples) {
- filtered_current_level_ = level_factor_ * filtered_current_level_ +
- (1 - level_factor_) * buffer_size_samples;
-
- // Account for time-scale operations (accelerate and pre-emptive expand) and
- // make sure that the filtered value remains non-negative.
+void BufferLevelFilter::Update(size_t buffer_size_packets,
+ int time_stretched_samples,
+ size_t packet_len_samples) {
+ // Filter:
+ // |filtered_current_level_| = |level_factor_| * |filtered_current_level_| +
+ // (1 - |level_factor_|) * |buffer_size_packets|
+ // |level_factor_| and |filtered_current_level_| are in Q8.
+ // |buffer_size_packets| is in Q0.
filtered_current_level_ =
- std::max(0.0, filtered_current_level_ - time_stretched_samples);
+ ((level_factor_ * filtered_current_level_) >> 8) +
+ ((256 - level_factor_) * rtc::dchecked_cast<int>(buffer_size_packets));
+
+ // Account for time-scale operations (accelerate and pre-emptive expand).
+ if (time_stretched_samples && packet_len_samples > 0) {
+ // Time-scaling has been performed since last filter update. Subtract the
+ // value of |time_stretched_samples| from |filtered_current_level_| after
+ // converting |time_stretched_samples| from samples to packets in Q8.
+ // Make sure that the filtered value remains non-negative.
+
+ int64_t time_stretched_packets =
+ (int64_t{time_stretched_samples} * (1 << 8)) /
+ rtc::dchecked_cast<int64_t>(packet_len_samples);
+
+ filtered_current_level_ = rtc::saturated_cast<int>(
+ std::max<int64_t>(0, filtered_current_level_ - time_stretched_packets));
+ }
}
-void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level_packets) {
- if (target_buffer_level_packets <= 1) {
- level_factor_ = 0.980;
- } else if (target_buffer_level_packets <= 3) {
- level_factor_ = 0.984;
- } else if (target_buffer_level_packets <= 7) {
- level_factor_ = 0.988;
+void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) {
+ if (target_buffer_level <= 1) {
+ level_factor_ = 251;
+ } else if (target_buffer_level <= 3) {
+ level_factor_ = 252;
+ } else if (target_buffer_level <= 7) {
+ level_factor_ = 253;
} else {
- level_factor_ = 0.992;
+ level_factor_ = 254;
}
}
+int BufferLevelFilter::filtered_current_level() const {
+ return filtered_current_level_;
+}
+
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/buffer_level_filter.h b/modules/audio_coding/neteq/buffer_level_filter.h
index 18b43a3..83388fb 100644
--- a/modules/audio_coding/neteq/buffer_level_filter.h
+++ b/modules/audio_coding/neteq/buffer_level_filter.h
@@ -24,24 +24,24 @@
virtual void Reset();
// Updates the filter. Current buffer size is |buffer_size_packets| (Q0).
- // |time_stretched_samples| is subtracted from the filtered value (thus
- // bypassing the filter operation).
- virtual void Update(size_t buffer_size_samples, int time_stretched_samples);
+ // If |time_stretched_samples| is non-zero, the value is converted to the
+ // corresponding number of packets, and is subtracted from the filtered
+ // value (thus bypassing the filter operation). |packet_len_samples| is the
+ // number of audio samples carried in each incoming packet.
+ virtual void Update(size_t buffer_size_packets,
+ int time_stretched_samples,
+ size_t packet_len_samples);
- // Set the current target buffer level in number of packets (obtained from
+ // Set the current target buffer level (obtained from
// DelayManager::base_target_level()). Used to select the appropriate
// filter coefficient.
- virtual void SetTargetBufferLevel(int target_buffer_level_packets);
+ virtual void SetTargetBufferLevel(int target_buffer_level);
- // Returns filtered current level in number of samples.
- virtual int filtered_current_level() const {
- // Round to nearest whole sample.
- return static_cast<int>(filtered_current_level_ + 0.5);
- }
+ virtual int filtered_current_level() const;
private:
- double level_factor_; // Filter factor for the buffer level filter.
- double filtered_current_level_;
+ int level_factor_; // Filter factor for the buffer level filter in Q8.
+ int filtered_current_level_; // Filtered current buffer level in Q8.
RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
};
diff --git a/modules/audio_coding/neteq/buffer_level_filter_unittest.cc b/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
index 0d4a0f2..1f12e73 100644
--- a/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
+++ b/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
@@ -30,22 +30,23 @@
for (int times = 10; times <= 50; times += 10) {
for (int value = 100; value <= 200; value += 10) {
filter.Reset();
- filter.SetTargetBufferLevel(1); // Makes filter coefficient 0.98.
+ filter.SetTargetBufferLevel(1); // Makes filter coefficient 251/256.
rtc::StringBuilder ss;
ss << "times = " << times << ", value = " << value;
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
for (int i = 0; i < times; ++i) {
- filter.Update(value, 0 /* time_stretched_samples */);
+ filter.Update(value, 0 /* time_stretched_samples */,
+ 160 /* packet_len_samples */);
}
// Expect the filtered value to be (theoretically)
- // (1 - 0.980 ^ |times|) * |value|.
- double expected_value_double = (1 - pow(0.98, times)) * value;
+ // (1 - (251/256) ^ |times|) * |value|.
+ double expected_value_double = (1 - pow(251.0 / 256.0, times)) * value;
int expected_value = static_cast<int>(expected_value_double);
// filtered_current_level() returns the value in Q8.
// The actual value may differ slightly from the expected value due to
// intermediate-stage rounding errors in the filter implementation.
// This is why we have to use EXPECT_NEAR with a tolerance of +/-1.
- EXPECT_NEAR(expected_value, filter.filtered_current_level(), 1);
+ EXPECT_NEAR(expected_value, filter.filtered_current_level() >> 8, 1);
}
}
}
@@ -57,60 +58,104 @@
const int kTimes = 10;
const int kValue = 100;
- filter.SetTargetBufferLevel(3); // Makes filter coefficient 0.984.
+ filter.SetTargetBufferLevel(3); // Makes filter coefficient 252/256.
for (int i = 0; i < kTimes; ++i) {
- filter.Update(kValue, 0 /* time_stretched_samples */);
+ filter.Update(kValue, 0 /* time_stretched_samples */,
+ 160 /* packet_len_samples */);
}
// Expect the filtered value to be
- // (1 - 0.984 ^ |kTimes|) * |kValue|.
- int expected_value = 15;
- EXPECT_EQ(expected_value, filter.filtered_current_level());
+ // (1 - (252/256) ^ |kTimes|) * |kValue|.
+ int expected_value = 14;
+ // filtered_current_level() returns the value in Q8.
+ EXPECT_EQ(expected_value, filter.filtered_current_level() >> 8);
filter.Reset();
- filter.SetTargetBufferLevel(7); // Makes filter coefficient 0.988.
+ filter.SetTargetBufferLevel(7); // Makes filter coefficient 253/256.
for (int i = 0; i < kTimes; ++i) {
- filter.Update(kValue, 0 /* time_stretched_samples */);
+ filter.Update(kValue, 0 /* time_stretched_samples */,
+ 160 /* packet_len_samples */);
}
// Expect the filtered value to be
- // (1 - 0.988 ^ |kTimes|) * |kValue|.
+ // (1 - (253/256) ^ |kTimes|) * |kValue|.
expected_value = 11;
- EXPECT_EQ(expected_value, filter.filtered_current_level());
+ // filtered_current_level() returns the value in Q8.
+ EXPECT_EQ(expected_value, filter.filtered_current_level() >> 8);
filter.Reset();
- filter.SetTargetBufferLevel(8); // Makes filter coefficient 0.992.
+ filter.SetTargetBufferLevel(8); // Makes filter coefficient 254/256.
for (int i = 0; i < kTimes; ++i) {
- filter.Update(kValue, 0 /* time_stretched_samples */);
+ filter.Update(kValue, 0 /* time_stretched_samples */,
+ 160 /* packet_len_samples */);
}
// Expect the filtered value to be
- // (1 - 0.992 ^ |kTimes|) * |kValue|.
- expected_value = 8;
- EXPECT_EQ(expected_value, filter.filtered_current_level());
+ // (1 - (254/256) ^ |kTimes|) * |kValue|.
+ expected_value = 7;
+ // filtered_current_level() returns the value in Q8.
+ EXPECT_EQ(expected_value, filter.filtered_current_level() >> 8);
}
TEST(BufferLevelFilter, TimeStretchedSamples) {
BufferLevelFilter filter;
- filter.SetTargetBufferLevel(1); // Makes filter coefficient 0.98.
+ filter.SetTargetBufferLevel(1); // Makes filter coefficient 251/256.
// Update 10 times with value 100.
const int kTimes = 10;
const int kValue = 100;
- const int kTimeStretchedSamples = 3;
+ const int kPacketSizeSamples = 160;
+ const int kNumPacketsStretched = 2;
+ const int kTimeStretchedSamples = kNumPacketsStretched * kPacketSizeSamples;
for (int i = 0; i < kTimes; ++i) {
- filter.Update(kValue, 0);
+ // Packet size set to 0. Do not expect the parameter
+ // |kTimeStretchedSamples| to have any effect.
+ filter.Update(kValue, kTimeStretchedSamples, 0 /* packet_len_samples */);
}
// Expect the filtered value to be
- // (1 - 0.98 ^ |kTimes|) * |kValue|.
- const int kExpectedValue = 18;
- EXPECT_EQ(kExpectedValue, filter.filtered_current_level());
+ // (1 - (251/256) ^ |kTimes|) * |kValue|.
+ const int kExpectedValue = 17;
+ // filtered_current_level() returns the value in Q8.
+ EXPECT_EQ(kExpectedValue, filter.filtered_current_level() >> 8);
// Update filter again, now with non-zero value for packet length.
// Set the current filtered value to be the input, in order to isolate the
// impact of |kTimeStretchedSamples|.
- filter.Update(filter.filtered_current_level(), kTimeStretchedSamples);
- EXPECT_EQ(kExpectedValue - kTimeStretchedSamples,
- filter.filtered_current_level());
+ filter.Update(filter.filtered_current_level() >> 8, kTimeStretchedSamples,
+ kPacketSizeSamples);
+ EXPECT_EQ(kExpectedValue - kNumPacketsStretched,
+ filter.filtered_current_level() >> 8);
// Try negative value and verify that we come back to the previous result.
- filter.Update(filter.filtered_current_level(), -kTimeStretchedSamples);
- EXPECT_EQ(kExpectedValue, filter.filtered_current_level());
+ filter.Update(filter.filtered_current_level() >> 8, -kTimeStretchedSamples,
+ kPacketSizeSamples);
+ EXPECT_EQ(kExpectedValue, filter.filtered_current_level() >> 8);
+}
+
+TEST(BufferLevelFilter, TimeStretchedSamplesNegativeUnevenFrames) {
+ BufferLevelFilter filter;
+ filter.SetTargetBufferLevel(1); // Makes filter coefficient 251/256.
+ // Update 10 times with value 100.
+ const int kTimes = 10;
+ const int kValue = 100;
+ const int kPacketSizeSamples = 160;
+ const int kTimeStretchedSamples = -3.1415 * kPacketSizeSamples;
+ for (int i = 0; i < kTimes; ++i) {
+ // Packet size set to 0. Do not expect the parameter
+ // |kTimeStretchedSamples| to have any effect.
+ filter.Update(kValue, kTimeStretchedSamples, 0 /* packet_len_samples */);
+ }
+ // Expect the filtered value to be
+ // (1 - (251/256) ^ |kTimes|) * |kValue|.
+ const int kExpectedValue = 17;
+ // filtered_current_level() returns the value in Q8.
+ EXPECT_EQ(kExpectedValue, filter.filtered_current_level() >> 8);
+
+ // Update filter again, now with non-zero value for packet length.
+ // Set the current filtered value to be the input, in order to isolate the
+ // impact of |kTimeStretchedSamples|.
+ filter.Update(filter.filtered_current_level() >> 8, kTimeStretchedSamples,
+ kPacketSizeSamples);
+ EXPECT_EQ(21, filter.filtered_current_level() >> 8);
+ // Try negative value and verify that we come back to the previous result.
+ filter.Update(filter.filtered_current_level() >> 8, -kTimeStretchedSamples,
+ kPacketSizeSamples);
+ EXPECT_EQ(kExpectedValue, filter.filtered_current_level() >> 8);
}
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/decision_logic.cc b/modules/audio_coding/neteq/decision_logic.cc
index f9f420a..40e421d 100644
--- a/modules/audio_coding/neteq/decision_logic.cc
+++ b/modules/audio_coding/neteq/decision_logic.cc
@@ -113,9 +113,11 @@
cng_state_ = kCngInternalOn;
}
+ const size_t samples_left =
+ sync_buffer.FutureLength() - expand.overlap_length();
// TODO(jakobi): Use buffer span instead of num samples.
const size_t cur_size_samples =
- packet_buffer_.NumSamplesInBuffer(decoder_frame_length);
+ samples_left + packet_buffer_.NumSamplesInBuffer(decoder_frame_length);
prev_time_scale_ =
prev_time_scale_ && (prev_mode == kModeAccelerateSuccess ||
@@ -173,7 +175,8 @@
// if the mute factor is low enough (otherwise the expansion was short enough
// to not be noticable).
// Note that the MuteFactor is in Q14, so a value of 16384 corresponds to 1.
- size_t current_span = packet_buffer_.GetSpanSamples(decoder_frame_length);
+ size_t current_span =
+ samples_left + packet_buffer_.GetSpanSamples(decoder_frame_length);
if ((prev_mode == kModeExpand || prev_mode == kModeCodecPlc) &&
expand.MuteFactor(0) < 16384 / 2 &&
current_span < static_cast<size_t>(delay_manager_->TargetLevel() *
@@ -190,9 +193,9 @@
return ExpectedPacketAvailable(prev_mode, play_dtmf);
} else if (!PacketBuffer::IsObsoleteTimestamp(
available_timestamp, target_timestamp, five_seconds_samples)) {
- return FuturePacketAvailable(decoder_frame_length, prev_mode,
- target_timestamp, available_timestamp,
- play_dtmf, generated_noise_samples);
+ return FuturePacketAvailable(
+ sync_buffer, expand, decoder_frame_length, prev_mode, target_timestamp,
+ available_timestamp, play_dtmf, generated_noise_samples);
} else {
// This implies that available_timestamp < target_timestamp, which can
// happen when a new stream or codec is received. Signal for a reset.
@@ -212,13 +215,19 @@
buffer_level_filter_->SetTargetBufferLevel(
delay_manager_->base_target_level());
+ size_t buffer_size_packets = 0;
+ if (packet_length_samples_ > 0) {
+ // Calculate size in packets.
+ buffer_size_packets = buffer_size_samples / packet_length_samples_;
+ }
int sample_memory_local = 0;
if (prev_time_scale_) {
sample_memory_local = sample_memory_;
timescale_countdown_ = tick_timer_->GetNewCountdown(kMinTimescaleInterval);
}
- buffer_level_filter_->Update(buffer_size_samples, sample_memory_local);
+ buffer_level_filter_->Update(buffer_size_packets, sample_memory_local,
+ packet_length_samples_);
prev_time_scale_ = false;
}
@@ -274,22 +283,15 @@
Operations DecisionLogic::ExpectedPacketAvailable(Modes prev_mode,
bool play_dtmf) {
if (!disallow_time_stretching_ && prev_mode != kModeExpand && !play_dtmf) {
- // Check criterion for time-stretching. The values are in number of packets
- // in Q8.
+ // Check criterion for time-stretching.
int low_limit, high_limit;
delay_manager_->BufferLimits(&low_limit, &high_limit);
- int buffer_level_packets = 0;
- if (packet_length_samples_ > 0) {
- buffer_level_packets =
- ((1 << 8) * buffer_level_filter_->filtered_current_level()) /
- packet_length_samples_;
- }
- if (buffer_level_packets >= high_limit << 2)
+ if (buffer_level_filter_->filtered_current_level() >= high_limit << 2)
return kFastAccelerate;
if (TimescaleAllowed()) {
- if (buffer_level_packets >= high_limit)
+ if (buffer_level_filter_->filtered_current_level() >= high_limit)
return kAccelerate;
- if (buffer_level_packets < low_limit)
+ if (buffer_level_filter_->filtered_current_level() < low_limit)
return kPreemptiveExpand;
}
}
@@ -297,6 +299,8 @@
}
Operations DecisionLogic::FuturePacketAvailable(
+ const SyncBuffer& sync_buffer,
+ const Expand& expand,
size_t decoder_frame_length,
Modes prev_mode,
uint32_t target_timestamp,
@@ -323,8 +327,10 @@
return kNormal;
}
+ const size_t samples_left =
+ sync_buffer.FutureLength() - expand.overlap_length();
const size_t cur_size_samples =
- packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
+ samples_left + packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
// If previous was comfort noise, then no merge is needed.
if (prev_mode == kModeRfc3389Cng || prev_mode == kModeCodecInternalCng) {
@@ -359,13 +365,8 @@
}
bool DecisionLogic::UnderTargetLevel() const {
- int buffer_level_packets = 0;
- if (packet_length_samples_ > 0) {
- buffer_level_packets =
- ((1 << 8) * buffer_level_filter_->filtered_current_level()) /
- packet_length_samples_;
- }
- return buffer_level_packets <= delay_manager_->TargetLevel();
+ return buffer_level_filter_->filtered_current_level() <=
+ delay_manager_->TargetLevel();
}
bool DecisionLogic::ReinitAfterExpands(uint32_t timestamp_leap) const {
diff --git a/modules/audio_coding/neteq/decision_logic.h b/modules/audio_coding/neteq/decision_logic.h
index 49020b0..2414e8c 100644
--- a/modules/audio_coding/neteq/decision_logic.h
+++ b/modules/audio_coding/neteq/decision_logic.h
@@ -134,7 +134,9 @@
// Returns the operation to do given that the expected packet is not
// available, but a packet further into the future is at hand.
- Operations FuturePacketAvailable(size_t decoder_frame_length,
+ Operations FuturePacketAvailable(const SyncBuffer& sync_buffer,
+ const Expand& expand,
+ size_t decoder_frame_length,
Modes prev_mode,
uint32_t target_timestamp,
uint32_t available_timestamp,
diff --git a/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h b/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
index 031195c..bf9fd59 100644
--- a/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
+++ b/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
@@ -22,8 +22,10 @@
virtual ~MockBufferLevelFilter() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD0(Reset, void());
- MOCK_METHOD2(Update,
- void(size_t buffer_size_samples, int time_stretched_samples));
+ MOCK_METHOD3(Update,
+ void(size_t buffer_size_packets,
+ int time_stretched_samples,
+ size_t packet_len_samples));
MOCK_METHOD1(SetTargetBufferLevel, void(int target_buffer_level));
MOCK_CONST_METHOD0(filtered_current_level, int());
};
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 82ec18d..ad6becc 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -310,12 +310,18 @@
int NetEqImpl::FilteredCurrentDelayMs() const {
rtc::CritScope lock(&crit_sect_);
+ // Calculate the filtered packet buffer level in samples. The value from
+ // |buffer_level_filter_| is in number of packets, represented in Q8.
+ const size_t packet_buffer_samples =
+ (buffer_level_filter_->filtered_current_level() *
+ decoder_frame_length_) >>
+ 8;
// Sum up the filtered packet buffer level with the future length of the sync
- // buffer.
- const int delay_samples = buffer_level_filter_->filtered_current_level() +
- sync_buffer_->FutureLength();
+ // buffer, and divide the sum by the sample rate.
+ const size_t delay_samples =
+ packet_buffer_samples + sync_buffer_->FutureLength();
// The division below will truncate. The return value is in ms.
- return delay_samples / rtc::CheckedDivExact(fs_hz_, 1000);
+ return static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
}
int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index a864985..6c67ca8 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -458,16 +458,16 @@
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
const std::string output_checksum =
- PlatformChecksum("415cc2366810676aad27a92b7a9068e79b7e4093",
- "52411c46e946e1d50fa777bbfb4686e7161f9cc7", "not used",
- "415cc2366810676aad27a92b7a9068e79b7e4093",
- "3e02f268f3b4f3acfcc1c397214ea13d09c212fd");
+ PlatformChecksum("9652cee1d6771a9cbfda821ae1bbdb41b0dd4dee",
+ "54a7e32f163663c0af35bf70bf45cefc24ad62ef", "not used",
+ "9652cee1d6771a9cbfda821ae1bbdb41b0dd4dee",
+ "79496b0a1ef0a3824f3ee04789748a461bed643f");
const std::string network_stats_checksum =
- PlatformChecksum("6c8726e8699cc61535e65ef9406d339954a05cf8",
- "b9f2c862328ddb8d3ec9e79f61cc1d937ee56df6", "not used",
- "6c8726e8699cc61535e65ef9406d339954a05cf8",
- "6c8726e8699cc61535e65ef9406d339954a05cf8");
+ PlatformChecksum("c59b1f9f282b6d8733cdff975e3c150ca4a47d51",
+ "bca95e565996a4ffd6e2ac15736e08843bdca93b", "not used",
+ "c59b1f9f282b6d8733cdff975e3c150ca4a47d51",
+ "c59b1f9f282b6d8733cdff975e3c150ca4a47d51");
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
FLAG_gen_ref);
@@ -486,17 +486,17 @@
// Checksum depends on libopus being compiled with or without SSE.
const std::string maybe_sse =
"14a63b3c7b925c82296be4bafc71bec85f2915c2|"
- "8884ffe98700454875a88d1c3ea9b55e829a2016";
+ "2c05677daa968d6c68b92adf4affb7cd9bb4d363";
const std::string output_checksum = PlatformChecksum(
- maybe_sse, "1f76c884d16812704dec85f2b98db697108d2d9b",
- "f888efef2d0710f202f19ba167ed509609146929", maybe_sse, maybe_sse);
+ maybe_sse, "b7b7ed802b0e18ee416973bf3b9ae98599b0181d",
+ "5876e52dda90d5ca433c3726555b907b97c86374", maybe_sse, maybe_sse);
const std::string network_stats_checksum =
- PlatformChecksum("bb8d91f71873df562224fe0dc5822b5dfb030090",
- "9f858290a675c4ceabf575feb6a59bef76d62eaf",
- "12963c965d9d102ebbc1f619508e2960a1f7dbee",
- "bb8d91f71873df562224fe0dc5822b5dfb030090",
- "bb8d91f71873df562224fe0dc5822b5dfb030090");
+ PlatformChecksum("adb3272498e436d1c019cbfd71610e9510c54497",
+ "fa935a91abc7291db47428a2d7c5361b98713a92",
+ "42106aa5267300f709f63737707ef07afd9dac61",
+ "adb3272498e436d1c019cbfd71610e9510c54497",
+ "adb3272498e436d1c019cbfd71610e9510c54497");
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
FLAG_gen_ref);
@@ -796,7 +796,7 @@
const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
const double kNetworkFreezeTimeMs = 5000.0;
const bool kGetAudioDuringFreezeRecovery = false;
- const int kDelayToleranceMs = 60;
+ const int kDelayToleranceMs = 50;
const int kMaxTimeToSpeechMs = 200;
LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
