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Unified Diff: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc

Issue 2546493002: Update smoothed bitrate. (Closed)
Patch Set: Changed cl according to offline discussions. Created 4 years ago
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« webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc ('K') | « webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc ('k') | webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h » ('j') | webrtc/voice_engine/channel.cc » ('J')
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Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index e8bfaf6830cca7dff72f7ef749b9b46ad8173634..89a9a40348797c47a435d512fd00a9ee30b8e553 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -11,6 +11,7 @@
#include <memory>
#include "webrtc/base/checks.h"
+#include "webrtc/common_audio/mocks/mock_smoothing_filter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
@@ -42,8 +43,10 @@ AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
struct AudioEncoderOpusStates {
std::shared_ptr<MockAudioNetworkAdaptor*> mock_audio_network_adaptor;
+ MockSmoothingFilter* mock_bitrate_smoother;
std::unique_ptr<AudioEncoderOpus> encoder;
std::unique_ptr<SimulatedClock> simulated_clock;
+ AudioEncoderOpus::Config config;
};
AudioEncoderOpusStates CreateCodec(size_t num_channels) {
@@ -68,11 +71,15 @@ AudioEncoderOpusStates CreateCodec(size_t num_channels) {
CodecInst codec_inst = kDefaultOpusSettings;
codec_inst.channels = num_channels;
- auto config = CreateConfig(codec_inst);
+ states.config = CreateConfig(codec_inst);
+ std::unique_ptr<MockSmoothingFilter> bitrate_smoother(
+ new MockSmoothingFilter());
+ states.mock_bitrate_smoother = bitrate_smoother.get();
states.simulated_clock.reset(new SimulatedClock(kInitialTimeUs));
- config.clock = states.simulated_clock.get();
+ states.config.clock = states.simulated_clock.get();
- states.encoder.reset(new AudioEncoderOpus(config, std::move(creator)));
+ states.encoder.reset(new AudioEncoderOpus(states.config, std::move(creator),
+ std::move(bitrate_smoother)));
return states;
}
@@ -159,20 +166,25 @@ TEST(AudioEncoderOpusTest,
const int kMinBitrateBps = 500;
const int kMaxBitrateBps = 512000;
// Set a too low bitrate.
- states.encoder->OnReceivedTargetAudioBitrate(kMinBitrateBps - 1);
+ states.encoder->OnReceivedTargetAudioBitrate(kMinBitrateBps - 1,
+ rtc::Optional<int64_t>());
EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
// Set a too high bitrate.
- states.encoder->OnReceivedTargetAudioBitrate(kMaxBitrateBps + 1);
+ states.encoder->OnReceivedTargetAudioBitrate(kMaxBitrateBps + 1,
+ rtc::Optional<int64_t>());
EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
// Set the minimum rate.
- states.encoder->OnReceivedTargetAudioBitrate(kMinBitrateBps);
+ states.encoder->OnReceivedTargetAudioBitrate(kMinBitrateBps,
+ rtc::Optional<int64_t>());
EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
// Set the maximum rate.
- states.encoder->OnReceivedTargetAudioBitrate(kMaxBitrateBps);
+ states.encoder->OnReceivedTargetAudioBitrate(kMaxBitrateBps,
+ rtc::Optional<int64_t>());
EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
// Set rates from 1000 up to 32000 bps.
for (int rate = 1000; rate <= 32000; rate += 1000) {
- states.encoder->OnReceivedTargetAudioBitrate(rate);
+ states.encoder->OnReceivedTargetAudioBitrate(rate,
+ rtc::Optional<int64_t>());
EXPECT_EQ(rate, states.encoder->GetTargetBitrate());
}
}
@@ -250,23 +262,6 @@ TEST(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
EXPECT_THAT(states.encoder->supported_frame_lengths_ms(), ElementsAre(20));
}
-TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
- auto states = CreateCodec(2);
- states.encoder->EnableAudioNetworkAdaptor("", nullptr);
-
- auto config = CreateEncoderRuntimeConfig();
- EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
- .WillOnce(Return(config));
-
- // Since using mock audio network adaptor, any bandwidth value is fine.
- constexpr int kUplinkBandwidth = 50000;
- EXPECT_CALL(**states.mock_audio_network_adaptor,
- SetUplinkBandwidth(kUplinkBandwidth));
- states.encoder->OnReceivedUplinkBandwidth(kUplinkBandwidth);
-
- CheckEncoderRuntimeConfig(states.encoder.get(), config);
-}
-
TEST(AudioEncoderOpusTest,
InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
auto states = CreateCodec(2);
@@ -296,9 +291,14 @@ TEST(AudioEncoderOpusTest,
// Since using mock audio network adaptor, any target audio bitrate is fine.
constexpr int kTargetAudioBitrate = 30000;
+ constexpr int64_t kProbingIntervalMs = 3000;
EXPECT_CALL(**states.mock_audio_network_adaptor,
SetTargetAudioBitrate(kTargetAudioBitrate));
- states.encoder->OnReceivedTargetAudioBitrate(kTargetAudioBitrate);
+ EXPECT_CALL(*states.mock_bitrate_smoother,
+ SetTimeConstantMs(kProbingIntervalMs * 4));
+ EXPECT_CALL(*states.mock_bitrate_smoother, AddSample(kTargetAudioBitrate));
+ states.encoder->OnReceivedTargetAudioBitrate(
+ kTargetAudioBitrate, rtc::Optional<int64_t>(kProbingIntervalMs));
CheckEncoderRuntimeConfig(states.encoder.get(), config);
}
@@ -367,7 +367,8 @@ TEST(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
auto states = CreateCodec(2);
- states.encoder->OnReceivedTargetAudioBitrate(kDefaultOpusSettings.rate * 2);
+ states.encoder->OnReceivedTargetAudioBitrate(kDefaultOpusSettings.rate * 2,
+ rtc::Optional<int64_t>());
// Since |OnReceivedOverhead| has not been called, the codec bitrate should
// not change.
@@ -384,7 +385,8 @@ TEST(AudioEncoderOpusTest, OverheadRemovedFromTargetAudioBitrate) {
states.encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
constexpr int kTargetBitrateBps = 40000;
- states.encoder->OnReceivedTargetAudioBitrate(kTargetBitrateBps);
+ states.encoder->OnReceivedTargetAudioBitrate(kTargetBitrateBps,
+ rtc::Optional<int64_t>());
int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
EXPECT_EQ(kTargetBitrateBps -
@@ -410,14 +412,16 @@ TEST(AudioEncoderOpusTest, BitrateBounded) {
// subtracted. The eventual codec rate should be bounded by |kMinBitrateBps|.
int target_bitrate =
kOverheadBytesPerPacket * 8 * packet_rate + kMinBitrateBps - 1;
- states.encoder->OnReceivedTargetAudioBitrate(target_bitrate);
+ states.encoder->OnReceivedTargetAudioBitrate(target_bitrate,
+ rtc::Optional<int64_t>());
EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
// Set a target rate that is greater than |kMaxBitrateBps| when overhead is
// subtracted. The eventual codec rate should be bounded by |kMaxBitrateBps|.
target_bitrate =
kOverheadBytesPerPacket * 8 * packet_rate + kMaxBitrateBps + 1;
- states.encoder->OnReceivedTargetAudioBitrate(target_bitrate);
+ states.encoder->OnReceivedTargetAudioBitrate(target_bitrate,
+ rtc::Optional<int64_t>());
EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
}
@@ -443,4 +447,33 @@ TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
config.bitrate_bps = rtc::Optional<int>(14001);
EXPECT_EQ(rtc::Optional<int>(6), config.GetNewComplexity());
}
+
+TEST(AudioEncoderOpusTest, UpdateUplinkBandwithInAudioNetworkAdaptor) {
minyue-webrtc 2016/12/22 14:51:27 Bandwidth
michaelt 2016/12/22 15:10:10 Done.
+ auto states = CreateCodec(2);
+ states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+ int16_t audio[480 * 2];
+ rtc::Buffer encoded;
+ EXPECT_CALL(*states.mock_bitrate_smoother, GetAverage())
+ .WillOnce(Return(rtc::Optional<float>(50000)));
+ EXPECT_CALL(**states.mock_audio_network_adaptor, SetUplinkBandwidth(50000));
+ states.encoder->Encode(0, rtc::ArrayView<const int16_t>(audio, 480 * 2),
+ &encoded);
+
+ // Repeat update uplink bandwidth tests.
+ for (int i = 0; i < 5; i++) {
+ // Don't update till its time to update again.
minyue-webrtc 2016/12/22 14:51:27 its -> it is
michaelt 2016/12/22 15:10:10 Done.
+ states.simulated_clock->AdvanceTimeMilliseconds(
+ states.config.update_uplink_bandwidth_interval_ms - 1);
+ states.encoder->Encode(0, rtc::ArrayView<const int16_t>(audio, 480 * 2),
+ &encoded);
+
+ // Update when its time to update.
minyue-webrtc 2016/12/22 14:51:27 its -> it is
michaelt 2016/12/22 15:10:10 Done.
+ EXPECT_CALL(*states.mock_bitrate_smoother, GetAverage())
+ .WillOnce(Return(rtc::Optional<float>(40000)));
+ EXPECT_CALL(**states.mock_audio_network_adaptor, SetUplinkBandwidth(40000));
+ states.simulated_clock->AdvanceTimeMilliseconds(1);
+ states.encoder->Encode(0, rtc::ArrayView<const int16_t>(audio, 480 * 2),
+ &encoded);
+ }
+}
} // namespace webrtc
« webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc ('K') | « webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc ('k') | webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h » ('j') | webrtc/voice_engine/channel.cc » ('J')

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