Injectable audio encoders: BuiltinAudioEncoderFactory

webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc

Index: webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
diff --git a/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..0d3ce5255ed110774210f8881efc9a9812f689d1
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
@@ -0,0 +1,145 @@
+/*
+ *  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 <limits>
+#include <memory>
+#include <vector>
+
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h"
+#include "webrtc/test/gmock.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(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, ensure they're within range.
+        audio[i] =
+            static_cast<int16_t>(i & std::numeric_limits<int16_t>::max());
+      }
+      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;
+      }
+    }
+    ASSERT_LT(blocks, kMaxEncodeCalls);
+  }
+}
+
+INSTANTIATE_TEST_CASE_P(BuiltinAudioEncoderFactoryTest,
+                        AudioEncoderFactoryTest,
+                        ::testing::Values(CreateBuiltinAudioEncoderFactory()));
+
+TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) {
+  using ::testing::ElementsAreArray;
+  // 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();
+
+  const std::vector<SdpAudioFormat> supported_formats = [&specs] {
+    std::vector<SdpAudioFormat> formats;
+    for (const auto& spec : specs) {
+      formats.push_back(spec.format);
+    }
+    return formats;
+  }();
+
+  const std::vector<SdpAudioFormat> expected_formats = {
+#ifdef WEBRTC_CODEC_OPUS
+    {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+    {"isac", 16000, 1},
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+    {"isac", 32000, 1},
+#endif
+#ifdef WEBRTC_CODEC_G722
+    {"G722", 8000, 1},
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+    {"ilbc", 8000, 1},
+#endif
+    {"pcmu", 8000, 1},
+    {"pcma", 8000, 1}
+  };
+
+  ASSERT_THAT(supported_formats, ElementsAreArray(expected_formats));
+}
+}  // namespace webrtc