Remove NetEq extra delay option.
Bug: b/156734419
Change-Id: I787e6961ad283990d633029c0cf296e10b825875
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237403
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35326}
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index 0c39d51..3e12006 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -11,8 +11,6 @@
#include "api/audio/audio_frame.h"
#include <string.h>
-#include <algorithm>
-#include <utility>
#include "rtc_base/checks.h"
#include "rtc_base/time_utils.h"
@@ -24,28 +22,6 @@
static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
}
-void swap(AudioFrame& a, AudioFrame& b) {
- using std::swap;
- swap(a.timestamp_, b.timestamp_);
- swap(a.elapsed_time_ms_, b.elapsed_time_ms_);
- swap(a.ntp_time_ms_, b.ntp_time_ms_);
- swap(a.samples_per_channel_, b.samples_per_channel_);
- swap(a.sample_rate_hz_, b.sample_rate_hz_);
- swap(a.num_channels_, b.num_channels_);
- swap(a.channel_layout_, b.channel_layout_);
- swap(a.speech_type_, b.speech_type_);
- swap(a.vad_activity_, b.vad_activity_);
- swap(a.profile_timestamp_ms_, b.profile_timestamp_ms_);
- swap(a.packet_infos_, b.packet_infos_);
- const size_t length_a = a.samples_per_channel_ * a.num_channels_;
- const size_t length_b = b.samples_per_channel_ * b.num_channels_;
- RTC_DCHECK_LE(length_a, AudioFrame::kMaxDataSizeSamples);
- RTC_DCHECK_LE(length_b, AudioFrame::kMaxDataSizeSamples);
- std::swap_ranges(a.data_, a.data_ + std::max(length_a, length_b), b.data_);
- swap(a.muted_, b.muted_);
- swap(a.absolute_capture_timestamp_ms_, b.absolute_capture_timestamp_ms_);
-}
-
void AudioFrame::Reset() {
ResetWithoutMuting();
muted_ = true;
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index 726b9a9..628a1ec 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -14,8 +14,6 @@
#include <stddef.h>
#include <stdint.h>
-#include <utility>
-
#include "api/audio/channel_layout.h"
#include "api/rtp_packet_infos.h"
#include "rtc_base/constructor_magic.h"
@@ -60,8 +58,6 @@
AudioFrame();
- friend void swap(AudioFrame& a, AudioFrame& b);
-
// Resets all members to their default state.
void Reset();
// Same as Reset(), but leaves mute state unchanged. Muting a frame requires
diff --git a/api/audio/test/audio_frame_unittest.cc b/api/audio/test/audio_frame_unittest.cc
index f8d3318..dbf45ce 100644
--- a/api/audio/test/audio_frame_unittest.cc
+++ b/api/audio/test/audio_frame_unittest.cc
@@ -133,54 +133,4 @@
EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
}
-TEST(AudioFrameTest, SwapFrames) {
- AudioFrame frame1, frame2;
- int16_t samples1[kNumChannelsMono * kSamplesPerChannel];
- for (size_t i = 0; i < kNumChannelsMono * kSamplesPerChannel; ++i) {
- samples1[i] = i;
- }
- frame1.UpdateFrame(kTimestamp, samples1, kSamplesPerChannel, kSampleRateHz,
- AudioFrame::kPLC, AudioFrame::kVadActive,
- kNumChannelsMono);
- frame1.set_absolute_capture_timestamp_ms(12345678);
- const auto frame1_channel_layout = frame1.channel_layout();
-
- int16_t samples2[(kNumChannelsMono + 1) * (kSamplesPerChannel + 1)];
- for (size_t i = 0; i < (kNumChannelsMono + 1) * (kSamplesPerChannel + 1);
- ++i) {
- samples2[i] = 1000 + i;
- }
- frame2.UpdateFrame(kTimestamp + 1, samples2, kSamplesPerChannel + 1,
- kSampleRateHz + 1, AudioFrame::kNormalSpeech,
- AudioFrame::kVadPassive, kNumChannelsMono + 1);
- const auto frame2_channel_layout = frame2.channel_layout();
-
- swap(frame1, frame2);
-
- EXPECT_EQ(kTimestamp + 1, frame1.timestamp_);
- ASSERT_EQ(kSamplesPerChannel + 1, frame1.samples_per_channel_);
- EXPECT_EQ(kSampleRateHz + 1, frame1.sample_rate_hz_);
- EXPECT_EQ(AudioFrame::kNormalSpeech, frame1.speech_type_);
- EXPECT_EQ(AudioFrame::kVadPassive, frame1.vad_activity_);
- ASSERT_EQ(kNumChannelsMono + 1, frame1.num_channels_);
- for (size_t i = 0; i < (kNumChannelsMono + 1) * (kSamplesPerChannel + 1);
- ++i) {
- EXPECT_EQ(samples2[i], frame1.data()[i]);
- }
- EXPECT_FALSE(frame1.absolute_capture_timestamp_ms());
- EXPECT_EQ(frame2_channel_layout, frame1.channel_layout());
-
- EXPECT_EQ(kTimestamp, frame2.timestamp_);
- ASSERT_EQ(kSamplesPerChannel, frame2.samples_per_channel_);
- EXPECT_EQ(kSampleRateHz, frame2.sample_rate_hz_);
- EXPECT_EQ(AudioFrame::kPLC, frame2.speech_type_);
- EXPECT_EQ(AudioFrame::kVadActive, frame2.vad_activity_);
- ASSERT_EQ(kNumChannelsMono, frame2.num_channels_);
- for (size_t i = 0; i < kNumChannelsMono * kSamplesPerChannel; ++i) {
- EXPECT_EQ(samples1[i], frame2.data()[i]);
- }
- EXPECT_EQ(12345678, frame2.absolute_capture_timestamp_ms());
- EXPECT_EQ(frame1_channel_layout, frame2.channel_layout());
-}
-
} // namespace webrtc
diff --git a/api/neteq/neteq.cc b/api/neteq/neteq.cc
index e8ef4db..155ddf2 100644
--- a/api/neteq/neteq.cc
+++ b/api/neteq/neteq.cc
@@ -30,8 +30,7 @@
<< ", min_delay_ms=" << min_delay_ms << ", enable_fast_accelerate="
<< (enable_fast_accelerate ? "true" : "false")
<< ", enable_muted_state=" << (enable_muted_state ? "true" : "false")
- << ", enable_rtx_handling=" << (enable_rtx_handling ? "true" : "false")
- << ", extra_output_delay_ms=" << extra_output_delay_ms;
+ << ", enable_rtx_handling=" << (enable_rtx_handling ? "true" : "false");
return ss.str();
}
diff --git a/api/neteq/neteq.h b/api/neteq/neteq.h
index dbfa071..031ad59 100644
--- a/api/neteq/neteq.h
+++ b/api/neteq/neteq.h
@@ -136,10 +136,6 @@
bool enable_rtx_handling = false;
absl::optional<AudioCodecPairId> codec_pair_id;
bool for_test_no_time_stretching = false; // Use only for testing.
- // Adds extra delay to the output of NetEq, without affecting jitter or
- // loss behavior. This is mainly for testing. Value must be a non-negative
- // multiple of 10 ms.
- int extra_output_delay_ms = 0;
};
enum ReturnCodes { kOK = 0, kFail = -1 };
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 3696427..6107b17 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -50,7 +50,6 @@
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
-#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
@@ -71,24 +70,6 @@
return controller_factory.CreateNetEqController(config);
}
-int GetDelayChainLengthMs(int config_extra_delay_ms) {
- constexpr char kExtraDelayFieldTrial[] = "WebRTC-Audio-NetEqExtraDelay";
- if (webrtc::field_trial::IsEnabled(kExtraDelayFieldTrial)) {
- const auto field_trial_string =
- webrtc::field_trial::FindFullName(kExtraDelayFieldTrial);
- int extra_delay_ms = -1;
- if (sscanf(field_trial_string.c_str(), "Enabled-%d", &extra_delay_ms) ==
- 1 &&
- extra_delay_ms >= 0 && extra_delay_ms <= 2000) {
- RTC_LOG(LS_INFO) << "Delay chain length set to " << extra_delay_ms
- << " ms in field trial";
- return (extra_delay_ms / 10) * 10; // Rounding down to multiple of 10.
- }
- }
- // Field trial not set, or invalid value read. Use value from config.
- return config_extra_delay_ms;
-}
-
} // namespace
NetEqImpl::Dependencies::Dependencies(
@@ -154,10 +135,7 @@
speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent",
10, // Report once every 10 s.
tick_timer_.get()),
- no_time_stretching_(config.for_test_no_time_stretching),
- output_delay_chain_ms_(
- GetDelayChainLengthMs(config.extra_output_delay_ms)),
- output_delay_chain_(rtc::CheckedDivExact(output_delay_chain_ms_, 10)) {
+ no_time_stretching_(config.for_test_no_time_stretching) {
RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
int fs = config.sample_rate_hz;
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
@@ -275,27 +253,8 @@
last_output_sample_rate_hz_ == 48000)
<< "Unexpected sample rate " << last_output_sample_rate_hz_;
- if (!output_delay_chain_.empty()) {
- if (output_delay_chain_empty_) {
- for (auto& f : output_delay_chain_) {
- f.CopyFrom(*audio_frame);
- }
- output_delay_chain_empty_ = false;
- delayed_last_output_sample_rate_hz_ = last_output_sample_rate_hz_;
- } else {
- RTC_DCHECK_GE(output_delay_chain_ix_, 0);
- RTC_DCHECK_LT(output_delay_chain_ix_, output_delay_chain_.size());
- swap(output_delay_chain_[output_delay_chain_ix_], *audio_frame);
- *muted = audio_frame->muted();
- output_delay_chain_ix_ =
- (output_delay_chain_ix_ + 1) % output_delay_chain_.size();
- delayed_last_output_sample_rate_hz_ = audio_frame->sample_rate_hz();
- }
- }
-
if (current_sample_rate_hz) {
- *current_sample_rate_hz = delayed_last_output_sample_rate_hz_.value_or(
- last_output_sample_rate_hz_);
+ *current_sample_rate_hz = last_output_sample_rate_hz_;
}
return kOK;
@@ -340,8 +299,7 @@
MutexLock lock(&mutex_);
if (delay_ms >= 0 && delay_ms <= 10000) {
RTC_DCHECK(controller_.get());
- return controller_->SetMinimumDelay(
- std::max(delay_ms - output_delay_chain_ms_, 0));
+ return controller_->SetMinimumDelay(delay_ms);
}
return false;
}
@@ -350,8 +308,7 @@
MutexLock lock(&mutex_);
if (delay_ms >= 0 && delay_ms <= 10000) {
RTC_DCHECK(controller_.get());
- return controller_->SetMaximumDelay(
- std::max(delay_ms - output_delay_chain_ms_, 0));
+ return controller_->SetMaximumDelay(delay_ms);
}
return false;
}
@@ -372,7 +329,7 @@
int NetEqImpl::TargetDelayMs() const {
MutexLock lock(&mutex_);
RTC_DCHECK(controller_.get());
- return controller_->TargetLevelMs() + output_delay_chain_ms_;
+ return controller_->TargetLevelMs();
}
int NetEqImpl::FilteredCurrentDelayMs() const {
@@ -382,8 +339,7 @@
const int delay_samples =
controller_->GetFilteredBufferLevel() + sync_buffer_->FutureLength();
// The division below will truncate. The return value is in ms.
- return delay_samples / rtc::CheckedDivExact(fs_hz_, 1000) +
- output_delay_chain_ms_;
+ return delay_samples / rtc::CheckedDivExact(fs_hz_, 1000);
}
int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
@@ -391,11 +347,6 @@
RTC_DCHECK(decoder_database_.get());
*stats = CurrentNetworkStatisticsInternal();
stats_->GetNetworkStatistics(decoder_frame_length_, stats);
- // Compensate for output delay chain.
- stats->mean_waiting_time_ms += output_delay_chain_ms_;
- stats->median_waiting_time_ms += output_delay_chain_ms_;
- stats->min_waiting_time_ms += output_delay_chain_ms_;
- stats->max_waiting_time_ms += output_delay_chain_ms_;
return 0;
}
@@ -417,10 +368,6 @@
RTC_DCHECK_GT(fs_hz_, 0);
stats.current_buffer_size_ms =
static_cast<uint16_t>(total_samples_in_buffers * 1000 / fs_hz_);
-
- // Compensate for output delay chain.
- stats.current_buffer_size_ms += output_delay_chain_ms_;
- stats.preferred_buffer_size_ms += output_delay_chain_ms_;
return stats;
}
@@ -464,19 +411,12 @@
// which is indicated by returning an empty value.
return absl::nullopt;
}
- size_t sum_samples_in_output_delay_chain = 0;
- for (const auto& audio_frame : output_delay_chain_) {
- sum_samples_in_output_delay_chain += audio_frame.samples_per_channel();
- }
- return timestamp_scaler_->ToExternal(
- playout_timestamp_ -
- static_cast<uint32_t>(sum_samples_in_output_delay_chain));
+ return timestamp_scaler_->ToExternal(playout_timestamp_);
}
int NetEqImpl::last_output_sample_rate_hz() const {
MutexLock lock(&mutex_);
- return delayed_last_output_sample_rate_hz_.value_or(
- last_output_sample_rate_hz_);
+ return last_output_sample_rate_hz_;
}
absl::optional<NetEq::DecoderFormat> NetEqImpl::GetDecoderFormat(
@@ -2072,9 +2012,8 @@
extracted_samples = packet->timestamp - first_timestamp + packet_duration;
RTC_DCHECK(controller_);
- stats_->JitterBufferDelay(
- packet_duration, waiting_time_ms + output_delay_chain_ms_,
- controller_->TargetLevelMs() + output_delay_chain_ms_);
+ stats_->JitterBufferDelay(packet_duration, waiting_time_ms,
+ controller_->TargetLevelMs());
packet_list->push_back(std::move(*packet)); // Store packet in list.
packet = absl::nullopt; // Ensure it's never used after the move.
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 98bf0e2..2522e31 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -399,22 +399,6 @@
ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(mutex_);
bool no_time_stretching_ RTC_GUARDED_BY(mutex_); // Only used for test.
rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(mutex_);
- // Data members used for adding extra delay to the output of NetEq.
- // The delay in ms (which is 10 times the number of elements in
- // output_delay_chain_).
- const int output_delay_chain_ms_ RTC_GUARDED_BY(mutex_);
- // Vector of AudioFrames which contains the delayed audio. Accessed as a
- // circular buffer.
- std::vector<AudioFrame> output_delay_chain_ RTC_GUARDED_BY(mutex_);
- // Index into output_delay_chain_.
- size_t output_delay_chain_ix_ RTC_GUARDED_BY(mutex_) = 0;
- // Did output_delay_chain_ get populated yet?
- bool output_delay_chain_empty_ RTC_GUARDED_BY(mutex_) = true;
- // Contains the sample rate of the AudioFrame last emitted from the delay
- // chain. If the extra output delay chain is not used, or if no audio has been
- // emitted yet, the variable is empty.
- absl::optional<int> delayed_last_output_sample_rate_hz_
- RTC_GUARDED_BY(mutex_);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index c55f6f1..c7e3f08 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -1105,186 +1105,5 @@
EXPECT_EQ(0, stats.preemptive_rate);
}
-namespace {
-// Helper classes and data types and functions for NetEqOutputDelayTest.
-
-class VectorAudioSink : public AudioSink {
- public:
- // Does not take ownership of the vector.
- VectorAudioSink(std::vector<int16_t>* output_vector) : v_(output_vector) {}
-
- virtual ~VectorAudioSink() = default;
-
- bool WriteArray(const int16_t* audio, size_t num_samples) override {
- v_->reserve(v_->size() + num_samples);
- for (size_t i = 0; i < num_samples; ++i) {
- v_->push_back(audio[i]);
- }
- return true;
- }
-
- private:
- std::vector<int16_t>* const v_;
-};
-
-struct TestResult {
- NetEqLifetimeStatistics lifetime_stats;
- NetEqNetworkStatistics network_stats;
- absl::optional<uint32_t> playout_timestamp;
- int target_delay_ms;
- int filtered_current_delay_ms;
- int sample_rate_hz;
-};
-
-// This class is used as callback object to NetEqTest to collect some stats
-// at the end of the simulation.
-class SimEndStatsCollector : public NetEqSimulationEndedCallback {
- public:
- SimEndStatsCollector(TestResult& result) : result_(result) {}
-
- void SimulationEnded(int64_t /*simulation_time_ms*/, NetEq* neteq) override {
- result_.playout_timestamp = neteq->GetPlayoutTimestamp();
- result_.target_delay_ms = neteq->TargetDelayMs();
- result_.filtered_current_delay_ms = neteq->FilteredCurrentDelayMs();
- result_.sample_rate_hz = neteq->last_output_sample_rate_hz();
- }
-
- private:
- TestResult& result_;
-};
-
-TestResult DelayLineNetEqTest(int delay_ms,
- std::vector<int16_t>* output_vector) {
- NetEq::Config config;
- config.for_test_no_time_stretching = true;
- config.extra_output_delay_ms = delay_ms;
- auto codecs = NetEqTest::StandardDecoderMap();
- NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
- {1, kRtpExtensionAudioLevel},
- {3, kRtpExtensionAbsoluteSendTime},
- {5, kRtpExtensionTransportSequenceNumber},
- {7, kRtpExtensionVideoContentType},
- {8, kRtpExtensionVideoTiming}};
- std::unique_ptr<NetEqInput> input = std::make_unique<NetEqRtpDumpInput>(
- webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"),
- rtp_ext_map, absl::nullopt /*No SSRC filter*/);
- std::unique_ptr<TimeLimitedNetEqInput> input_time_limit(
- new TimeLimitedNetEqInput(std::move(input), 10000));
- std::unique_ptr<AudioSink> output =
- std::make_unique<VectorAudioSink>(output_vector);
-
- TestResult result;
- SimEndStatsCollector stats_collector(result);
- NetEqTest::Callbacks callbacks;
- callbacks.simulation_ended_callback = &stats_collector;
-
- NetEqTest test(config, CreateBuiltinAudioDecoderFactory(), codecs,
- /*text_log=*/nullptr, /*neteq_factory=*/nullptr,
- /*input=*/std::move(input_time_limit), std::move(output),
- callbacks);
- test.Run();
- result.lifetime_stats = test.LifetimeStats();
- result.network_stats = test.SimulationStats();
- return result;
-}
-} // namespace
-
-// Tests the extra output delay functionality of NetEq.
-TEST(NetEqOutputDelayTest, RunTest) {
- std::vector<int16_t> output;
- const auto result_no_delay = DelayLineNetEqTest(0, &output);
- std::vector<int16_t> output_delayed;
- constexpr int kDelayMs = 100;
- const auto result_delay = DelayLineNetEqTest(kDelayMs, &output_delayed);
-
- // Verify that the loss concealment remains unchanged. The point of the delay
- // is to not affect the jitter buffering behavior.
- // First verify that there are concealments in the test.
- EXPECT_GT(result_no_delay.lifetime_stats.concealed_samples, 0u);
- // And that not all of the output is concealment.
- EXPECT_GT(result_no_delay.lifetime_stats.total_samples_received,
- result_no_delay.lifetime_stats.concealed_samples);
- // Now verify that they remain unchanged by the delay.
- EXPECT_EQ(result_no_delay.lifetime_stats.concealed_samples,
- result_delay.lifetime_stats.concealed_samples);
- // Accelerate and pre-emptive expand should also be unchanged.
- EXPECT_EQ(result_no_delay.lifetime_stats.inserted_samples_for_deceleration,
- result_delay.lifetime_stats.inserted_samples_for_deceleration);
- EXPECT_EQ(result_no_delay.lifetime_stats.removed_samples_for_acceleration,
- result_delay.lifetime_stats.removed_samples_for_acceleration);
- // Verify that delay stats are increased with the delay chain.
- EXPECT_EQ(
- result_no_delay.lifetime_stats.jitter_buffer_delay_ms +
- kDelayMs * result_no_delay.lifetime_stats.jitter_buffer_emitted_count,
- result_delay.lifetime_stats.jitter_buffer_delay_ms);
- EXPECT_EQ(
- result_no_delay.lifetime_stats.jitter_buffer_target_delay_ms +
- kDelayMs * result_no_delay.lifetime_stats.jitter_buffer_emitted_count,
- result_delay.lifetime_stats.jitter_buffer_target_delay_ms);
- EXPECT_EQ(result_no_delay.network_stats.current_buffer_size_ms + kDelayMs,
- result_delay.network_stats.current_buffer_size_ms);
- EXPECT_EQ(result_no_delay.network_stats.preferred_buffer_size_ms + kDelayMs,
- result_delay.network_stats.preferred_buffer_size_ms);
- EXPECT_EQ(result_no_delay.network_stats.mean_waiting_time_ms + kDelayMs,
- result_delay.network_stats.mean_waiting_time_ms);
- EXPECT_EQ(result_no_delay.network_stats.median_waiting_time_ms + kDelayMs,
- result_delay.network_stats.median_waiting_time_ms);
- EXPECT_EQ(result_no_delay.network_stats.min_waiting_time_ms + kDelayMs,
- result_delay.network_stats.min_waiting_time_ms);
- EXPECT_EQ(result_no_delay.network_stats.max_waiting_time_ms + kDelayMs,
- result_delay.network_stats.max_waiting_time_ms);
-
- ASSERT_TRUE(result_no_delay.playout_timestamp);
- ASSERT_TRUE(result_delay.playout_timestamp);
- EXPECT_EQ(*result_no_delay.playout_timestamp -
- static_cast<uint32_t>(
- kDelayMs *
- rtc::CheckedDivExact(result_no_delay.sample_rate_hz, 1000)),
- *result_delay.playout_timestamp);
- EXPECT_EQ(result_no_delay.target_delay_ms + kDelayMs,
- result_delay.target_delay_ms);
- EXPECT_EQ(result_no_delay.filtered_current_delay_ms + kDelayMs,
- result_delay.filtered_current_delay_ms);
-
- // Verify expected delay in decoded signal. The test vector uses 8 kHz sample
- // rate, so the delay will be 8 times the delay in ms.
- constexpr size_t kExpectedDelaySamples = kDelayMs * 8;
- for (size_t i = 0;
- i < output.size() && i + kExpectedDelaySamples < output_delayed.size();
- ++i) {
- EXPECT_EQ(output[i], output_delayed[i + kExpectedDelaySamples]);
- }
-}
-
-// Tests the extra output delay functionality of NetEq when configured via
-// field trial.
-TEST(NetEqOutputDelayTest, RunTestWithFieldTrial) {
- test::ScopedFieldTrials field_trial(
- "WebRTC-Audio-NetEqExtraDelay/Enabled-50/");
- constexpr int kExpectedDelayMs = 50;
- std::vector<int16_t> output;
- const auto result = DelayLineNetEqTest(0, &output);
-
- // The base delay values are taken from the resuts of the non-delayed case in
- // NetEqOutputDelayTest.RunTest above.
- EXPECT_EQ(20 + kExpectedDelayMs, result.target_delay_ms);
- EXPECT_EQ(60 + kExpectedDelayMs, result.filtered_current_delay_ms);
-}
-
-// Set a non-multiple-of-10 value in the field trial, and verify that we don't
-// crash, and that the result is rounded down.
-TEST(NetEqOutputDelayTest, RunTestWithFieldTrialOddValue) {
- test::ScopedFieldTrials field_trial(
- "WebRTC-Audio-NetEqExtraDelay/Enabled-103/");
- constexpr int kRoundedDelayMs = 100;
- std::vector<int16_t> output;
- const auto result = DelayLineNetEqTest(0, &output);
-
- // The base delay values are taken from the resuts of the non-delayed case in
- // NetEqOutputDelayTest.RunTest above.
- EXPECT_EQ(20 + kRoundedDelayMs, result.target_delay_ms);
- EXPECT_EQ(60 + kRoundedDelayMs, result.filtered_current_delay_ms);
-}
-
} // namespace test
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc b/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
index 337f54e..3f06b1c 100644
--- a/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
+++ b/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
@@ -33,8 +33,7 @@
stats_getter_.reset(new NetEqStatsGetter(std::move(delay_analyzer)));
}
-void NetEqStatsPlotter::SimulationEnded(int64_t simulation_time_ms,
- NetEq* /*neteq*/) {
+void NetEqStatsPlotter::SimulationEnded(int64_t simulation_time_ms) {
if (make_matlab_plot_) {
auto matlab_script_name = base_file_name_;
std::replace(matlab_script_name.begin(), matlab_script_name.end(), '.',
diff --git a/modules/audio_coding/neteq/tools/neteq_stats_plotter.h b/modules/audio_coding/neteq/tools/neteq_stats_plotter.h
index d691867..c4df24e 100644
--- a/modules/audio_coding/neteq/tools/neteq_stats_plotter.h
+++ b/modules/audio_coding/neteq/tools/neteq_stats_plotter.h
@@ -28,7 +28,7 @@
bool show_concealment_events,
std::string base_file_name);
- void SimulationEnded(int64_t simulation_time_ms, NetEq* neteq) override;
+ void SimulationEnded(int64_t simulation_time_ms) override;
NetEqStatsGetter* stats_getter() { return stats_getter_.get(); }
diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc
index 22f5ad6..7902438 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -91,8 +91,7 @@
simulation_time += step_result.simulation_step_ms;
} while (!step_result.is_simulation_finished);
if (callbacks_.simulation_ended_callback) {
- callbacks_.simulation_ended_callback->SimulationEnded(simulation_time,
- neteq_.get());
+ callbacks_.simulation_ended_callback->SimulationEnded(simulation_time);
}
return simulation_time;
}
diff --git a/modules/audio_coding/neteq/tools/neteq_test.h b/modules/audio_coding/neteq/tools/neteq_test.h
index 3b787a6..0a6c24f 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_test.h
@@ -61,7 +61,7 @@
class NetEqSimulationEndedCallback {
public:
virtual ~NetEqSimulationEndedCallback() = default;
- virtual void SimulationEnded(int64_t simulation_time_ms, NetEq* neteq) = 0;
+ virtual void SimulationEnded(int64_t simulation_time_ms) = 0;
};
// Class that provides an input--output test for NetEq. The input (both packets
