Add opus audio codec support in remoting

remoting/codec/audio_encoder_opus_unittest.cc

+// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// MSVC++ requires this to get M_PI.
+#define _USE_MATH_DEFINES
+#include <math.h>
+
+#include "remoting/codec/audio_encoder_opus.h"
+
+#include "base/logging.h"
+#include "remoting/codec/audio_decoder_opus.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace remoting {
+
+namespace {
+
+const int kChannels = 2;
+const double kChannelPhaseShift = 2 * M_PI / 3;

[Wez, 2012/10/22 22:50:21]

nit: Add comments explaining what this is used for.

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+const AudioPacket_SamplingRate kDefaultSamplingRate =
+    AudioPacket::SAMPLING_RATE_48000;
+const int kMaxLatencyMs = 40;
+
+// When verifying results ignore the first 1k samples. This is necessary because
+// it takes some time for the codec to adjust for the input signal.
+const int kSkippedFirstSamples = 1000;
+
+}  // namespace
+
+class OpusAudioEncoderTest : public testing::Test {
+ public:
+
+  static int16 GetSampleValue(
+      AudioPacket::SamplingRate rate,
+      double frequency,

[Wez, 2012/10/22 22:50:21]

nit: Specify units.

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+      double pos,

[Wez, 2012/10/22 22:50:21]

nit: How can we have sub-sample position?

[Sergey Ulanov, 2012/10/23 00:43:50]

That's used when comparing results - GetSignalShift() returns sub-sample
position.

+      int channel) {
+    double angle = pos * 2 * M_PI * frequency / rate +
+        kChannelPhaseShift * channel;
+    return static_cast<int>(sin(angle) * 32767 + 0.5);
+  }
+
+  scoped_ptr<AudioPacket> CreatePacket(
+      int samples,
+      AudioPacket::SamplingRate rate,
+      double frequency,
+      int pos) {
+
+    std::vector<int16> data(samples * kChannels);

[Wez, 2012/10/22 22:50:21]

nit: Add a comment to explain that we're filling the channels with sine wave
sample data with different phase-shifts.

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+    for (int i = 0; i < samples; ++i) {
+      data[i * kChannels] = GetSampleValue(rate, frequency, i + pos, 0);
+      data[i * kChannels + 1] = GetSampleValue(rate, frequency, i + pos, 1);
+    }
+
+    scoped_ptr<AudioPacket> packet(new AudioPacket());
+    packet->add_data(reinterpret_cast<char*>(&(data[0])),
+                     samples * kChannels * sizeof(int16));
+    packet->set_encoding(AudioPacket::ENCODING_RAW);
+    packet->set_sampling_rate(rate);
+    packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
+    packet->set_channels(AudioPacket::CHANNELS_STEREO);
+    return packet.Pass();
+  }
+
+  // Decoded data is normally shifted in phase relative to the raw date. This

[Wez, 2012/10/22 22:50:21]

typo: date -> data

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+  // function returns approximate the phase difference by finding the first

[Wez, 2012/10/22 22:50:21]

typo: approximate the phase ... -> ... the approximate phase ...

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+  // point when signal goes from negative to positive.
+  double GetSignalShift(const std::vector<int16>& received_data) {

[Wez, 2012/10/22 22:50:21]

nit: So this function is supposed to be getting the phase shift using a
heuristic -> call it GetPhaseShift or EstimatePhaseShift?

[Sergey Ulanov, 2012/10/23 00:43:50]

Renamed to EstimateSignalShift(). I don't think it's correct to call it phase
shift because it's measured in samples instead of radians (updated comments to
make it clear).

+    for (size_t i = kSkippedFirstSamples;
+         i < received_data.size() / kChannels - 1; i++) {
+      int16 this_sample = received_data[i * kChannels];
+      int16 next_sample = received_data[(i + 1) * kChannels];
+      if (this_sample < 0 && next_sample > 0) {
+        return i +
+            static_cast<double>(-this_sample) / (next_sample - this_sample);
+      }
+    }
+    return 0;
+  }
+
+  void ValidateReceivedData(int samples,
+                            AudioPacket::SamplingRate rate,
+                            double frequency,
+                            const std::vector<int16>& received_data) {

[Wez, 2012/10/22 22:50:21]

nit: Add a comment explaining that we validate the data by estimating the source
phase shift, generating data with that shift and calculating the difference
between that and the decoded version.

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+    double shift = GetSignalShift(received_data);
+    double diff_sqare_sum = 0;
+    for (size_t i = kSkippedFirstSamples;
+         i < received_data.size() / kChannels; i++) {
+      double d = received_data[i * kChannels] -
+          GetSampleValue(rate, frequency, i - shift, 0);
+      diff_sqare_sum += d * d;
+      d = received_data[i * kChannels + 1] -
+          GetSampleValue(rate, frequency, i - shift, 1);
+      diff_sqare_sum += d * d;
+    }
+    double averate_difference = sqrt(diff_sqare_sum / received_data.size());

[Wez, 2012/10/22 22:50:21]

typo: average_difference

[Wez, 2012/10/22 22:50:21]

nit: Clarify the units you're using for the difference, to explain the magic
value of 5000

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+    EXPECT_LE(averate_difference, 5000);

[Wez, 2012/10/22 22:50:21]

nit: Consider moving 5000 to a constant, above

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+  }
+
+  void ResetEncoder() {

[Wez, 2012/10/22 22:50:21]

nit: ResetEncoder -> Reset or ResetEncoderAndDecoder

[Sergey Ulanov, 2012/10/23 00:43:50]

Removed this function.

+    encoder_.reset(new AudioEncoderOpus());
+    decoder_.reset(new AudioDecoderOpus());
+  }
+
+  void TestEncoderDecoder(int packet_size,

[Wez, 2012/10/22 22:50:21]

typo: TestEncodeDecode

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+                          double frequency,
+                          AudioPacket::SamplingRate rate) {
+    const int kTotalTestSamples = 24000;
+    ResetEncoder();
+
+    std::vector<int16> received_data;
+    int pos = 0;
+    for (; pos < kTotalTestSamples; pos += packet_size) {
+        scoped_ptr<AudioPacket> source_packet =
+            CreatePacket(packet_size, rate, frequency, pos);
+        scoped_ptr<AudioPacket> encoded = encoder_->Encode(source_packet.Pass());

[Wez, 2012/10/22 22:50:21]

Line too long?

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+        if (encoded.get()) {

[Wez, 2012/10/22 22:50:21]

EXPECT(encoded.get())?

[Sergey Ulanov, 2012/10/23 00:43:50]

Encoder is allowed to return NULL packet sometimes, particularly when original
raw packet is smaller than the encoder frame size.

+          scoped_ptr<AudioPacket> decoded = decoder_->Decode(encoded.Pass());
+          EXPECT_EQ(kDefaultSamplingRate, decoded->sampling_rate());
+          for (int i = 0; i < decoded->data_size(); ++i) {
+            const int16* data =
+                reinterpret_cast<const int16*>(decoded->data(i).data());
+            received_data.insert(
+                received_data.end(), data,
+                data + decoded->data(i).size() / sizeof(int16));
+          }
+        }
+    }
+
+    // Verify that at most kMaxLatencyMs worth of samples is buffered inside
+    // |encoder_| and |decoder_|.
+    EXPECT_GE(static_cast<int>(received_data.size()) / kChannels,
+              pos - rate * kMaxLatencyMs / 1000);
+
+    ValidateReceivedData(packet_size, kDefaultSamplingRate,
+                         frequency, received_data);
+  }
+
+ protected:
+  scoped_ptr<AudioEncoderOpus> encoder_;
+  scoped_ptr<AudioDecoderOpus> decoder_;
+};
+
+TEST_F(OpusAudioEncoderTest, CreateAndDestroy) {
+}
+
+TEST_F(OpusAudioEncoderTest, NoResampling) {
+  TestEncoderDecoder(2000, 50, AudioPacket::SAMPLING_RATE_48000);
+  TestEncoderDecoder(2000, 3000, AudioPacket::SAMPLING_RATE_48000);
+  TestEncoderDecoder(2000, 10000, AudioPacket::SAMPLING_RATE_48000);
+}
+
+TEST_F(OpusAudioEncoderTest, Resampling) {
+  TestEncoderDecoder(2000, 50, AudioPacket::SAMPLING_RATE_44100);
+  TestEncoderDecoder(2000, 3000, AudioPacket::SAMPLING_RATE_44100);
+  TestEncoderDecoder(2000, 10000, AudioPacket::SAMPLING_RATE_44100);
+}
+
+TEST_F(OpusAudioEncoderTest, BufferSize) {

[Wez, 2012/10/22 22:50:21]

nit: BufferingAndResampling / BufferSizeAndResampling?

[Sergey Ulanov, 2012/10/23 00:43:50]

Done.

+  TestEncoderDecoder(500, 3000, AudioPacket::SAMPLING_RATE_44100);
+  TestEncoderDecoder(1000, 3000, AudioPacket::SAMPLING_RATE_44100);
+  TestEncoderDecoder(5000, 3000, AudioPacket::SAMPLING_RATE_44100);

[Wez, 2012/10/22 22:50:21]

How long to these tests take?

[Sergey Ulanov, 2012/10/23 00:43:50]

About 0.6 seconds total for all tests in this file with Debug build, i.e. about
60ms each small test.

+}
+
+}  // namespace remoting