Injectable audio encoders: BuiltinAudioEncoderFactory

webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc

+/*
+ *  Copyright (c) 2017 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 <memory>
+
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+class AudioEncoderFactoryTest
+    : public ::testing::TestWithParam<rtc::scoped_refptr<AudioEncoderFactory>> {
+};
+
+TEST_P(AudioEncoderFactoryTest, SupportsAtLeastOneFormat) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  EXPECT_FALSE(supported_encoders.empty());
+}
+
+TEST_P(AudioEncoderFactoryTest, CanQueryAllSupportedFormats) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto info = factory->QueryAudioEncoder(spec.format);
+    EXPECT_TRUE(info);
+  }
+}
+
+TEST_P(AudioEncoderFactoryTest, CanConstructAllSupportedEncoders) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto info = factory->QueryAudioEncoder(spec.format);
+    auto encoder = factory->MakeAudioEncoder(127, spec.format);
+    EXPECT_TRUE(encoder);
+    EXPECT_EQ(encoder->SampleRateHz(), info->sample_rate_hz);
+    EXPECT_EQ(static_cast<int>(encoder->NumChannels()), info->num_channels);
+    EXPECT_EQ(encoder->RtpTimestampRateHz(), spec.format.clockrate_hz);
+  }
+}
+
+TEST_P(AudioEncoderFactoryTest, CanRunAllSupportedEncoders) {
+  constexpr int kTestPayloadType = 127;
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto encoder = factory->MakeAudioEncoder(kTestPayloadType, spec.format);
+    EXPECT_TRUE(encoder);
+    encoder->Reset();
+    const int num_samples =
+        encoder->SampleRateHz() * encoder->NumChannels() / 100;
+    rtc::Buffer out;
+    rtc::BufferT<int16_t> audio;
+    audio.SetData(num_samples, [](rtc::ArrayView<int16_t> audio) {
+      for (size_t i = 0; i != audio.size(); ++i) {
+        // Just put some numbers in there, allow for wrap-around.
+        audio[i] = static_cast<int16_t>(i);

[kwiberg-webrtc, 2017/03/24 12:43:25]

Overflow when casting from unsigned to signed is implementation-defined, IIRC.
Consider using something like i % (std::numeric_limits<int16_t>::max() + 1)
instead.

[ossu, 2017/04/05 14:41:52]

Acknowledged.

+      }
+      return audio.size();
+    });
+    // This is here to stop the test going forever with a broken encoder.
+    constexpr int kMaxEncodeCalls = 100;
+    int blocks = 0;
+    for (; blocks < kMaxEncodeCalls; ++blocks) {
+      AudioEncoder::EncodedInfo info = encoder->Encode(
+          blocks * encoder->RtpTimestampRateHz() / 100, audio, &out);
+      EXPECT_EQ(info.encoded_bytes, out.size());
+      if (info.encoded_bytes > 0) {
+        EXPECT_EQ(0u, info.encoded_timestamp);
+        EXPECT_EQ(kTestPayloadType, info.payload_type);
+        break;
+      }
+    }
+    ASSERT_LT(blocks, kMaxEncodeCalls);
+    const unsigned int next_timestamp =
+        blocks * encoder->RtpTimestampRateHz() / 100;
+    out.Clear();
+    for (; blocks < kMaxEncodeCalls; ++blocks) {
+      AudioEncoder::EncodedInfo info = encoder->Encode(
+          blocks * encoder->RtpTimestampRateHz() / 100, audio, &out);
+      EXPECT_EQ(info.encoded_bytes, out.size());
+      if (info.encoded_bytes > 0) {
+        EXPECT_EQ(next_timestamp, info.encoded_timestamp);
+        EXPECT_EQ(kTestPayloadType, info.payload_type);
+        break;
+      }
+    }

[kwiberg-webrtc, 2017/03/24 12:43:25]

You never test that the encoder actually returns a packet, or did I miss
something?

[ossu, 2017/04/05 14:41:52]

I was thinking this check did just that, though the comment above doesn't
specifically mention it.
If info.encoded_bytes is never > 0, then we will fail on this ASSERT.

+    ASSERT_LT(blocks, kMaxEncodeCalls);
+  }
+}
+
+INSTANTIATE_TEST_CASE_P(
+    BuiltinAudioEncoderFactoryTest,
+    AudioEncoderFactoryTest,
+    ::testing::Values(CreateBuiltinAudioEncoderFactory()));
+
+TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) {
+  // Check that we claim to support the formats we expect from build flags, and
+  // we've ordered them correctly.
+  auto factory = CreateBuiltinAudioEncoderFactory();
+  auto specs = factory->GetSupportedEncoders();
+  auto iter = specs.begin();
+  auto next_is_format = [&] (const SdpAudioFormat& format) {
+    if (iter != specs.end()) {
+      const bool found = iter->format == format;
+      ++iter;
+      return found;
+    }
+    return false;
+  };
+#ifdef WEBRTC_CODEC_OPUS
+  ASSERT_TRUE(next_is_format(
+      {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}));
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+    ASSERT_TRUE(next_is_format({"isac", 16000, 1}));
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+    ASSERT_TRUE(next_is_format({"isac", 32000, 1}));
+#endif
+#ifdef WEBRTC_CODEC_G722
+    ASSERT_TRUE(next_is_format({"G722", 8000, 1}));
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+    ASSERT_TRUE(next_is_format({"ilbc", 8000, 1}));
+#endif
+  ASSERT_TRUE(next_is_format({"pcmu", 8000, 1}));
+  ASSERT_TRUE(next_is_format({"pcma", 8000, 1}));

[kwiberg-webrtc, 2017/03/24 12:43:25]

Can you use ASSERT_THAT with ElementsAre() or something like that?
https://github.com/google/googletest/blob/master/googlemock/docs/CheatSheet.m...

[ossu, 2017/04/05 14:41:52]

I can! And it gives a neat printout when it fails!

+}
+}  // namespace webrtc