Let Opus increase complexity for low bitrates

webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc

+/*
+ *  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/base/format_macros.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
+#include "webrtc/test/gtest.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/perf_test.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace {
+int64_t RunComplexityTest(const AudioEncoderOpus::Config& config) {
+  // Create encoder.
+  AudioEncoderOpus encoder(config);
+  // Open speech file.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
+  test::AudioLoop audio_loop;
+  constexpr int kSampleRateHz = 48000;
+  EXPECT_EQ(kSampleRateHz, encoder.SampleRateHz());
+  constexpr size_t kMaxLoopLengthSamples =
+      kSampleRateHz * 10;  // 10 second loop.
+  constexpr size_t kInputBlockSizeSamples =
+      10 * kSampleRateHz / 1000;  // 60 ms.
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                              kInputBlockSizeSamples));
+  // Encode.
+  webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
+  const int64_t start_time_ms = clock->TimeInMilliseconds();
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  uint32_t rtp_timestamp = 0u;
+  for (size_t i = 0; i < 10000; ++i) {
+    encoded.Clear();
+    info = encoder.Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
+    rtp_timestamp += kInputBlockSizeSamples;
+  }
+  return clock->TimeInMilliseconds() - start_time_ms;
+}
+}  // namespace
+
+// This test encodes an audio file using Opus twice with different bitrates
+// (12.5 kbps and 15.5 kbps). The runtime for each is measured, and the ratio
+// between the two is calculated and tracked. This test explicitly sets the
+// low_rate_complexity to 9. When running on desktop platforms, this is the same
+// as the regular complexity, and the expectation is that the resulting ratio
+// should be less than 100% (since the encoder runs faster at lower bitrates,
+// given a fixed complexity setting). On the other hand, when running on
+// mobiles, the regular complexity is 5, and we expect the resulting ration to

[minyue-webrtc, 2016/11/22 09:48:34]

ratio

[hlundin-webrtc, 2016/11/22 09:52:21]

Done.

+// be higher, since we have explicitly asked for a higher complexity setting at
+// the lower rate.
+TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) {
+  // Create config.
+  AudioEncoderOpus::Config config;
+  config.bitrate_bps = rtc::Optional<int>(12500);
+  config.low_rate_complexity = 9;
+  int64_t runtime_12500bps = RunComplexityTest(config);
+
+  config.bitrate_bps = rtc::Optional<int>(15500);
+  int64_t runtime_15500bps = RunComplexityTest(config);
+
+  test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_on",
+                    100.0 * runtime_12500bps / runtime_15500bps, "percent",
+                    true);
+}
+
+// This test is identical to the one above, but without the complexity
+// adaptation enabled (neither on desktop, nor on mobile). The expectation is
+// that the resulting ratio is less than 100% at all times.
+TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) {
+  // Create config.
+  AudioEncoderOpus::Config config;
+  config.bitrate_bps = rtc::Optional<int>(12500);
+  int64_t runtime_12500bps = RunComplexityTest(config);
+
+  config.bitrate_bps = rtc::Optional<int>(15500);
+  int64_t runtime_15500bps = RunComplexityTest(config);
+
+  test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off",
+                    100.0 * runtime_12500bps / runtime_15500bps, "", true);
+}
+}  // namespace webrtc