Injectable audio encoders: BuiltinAudioEncoderFactory

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc

Index: webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index c7bbfb0434428a75dd6c63462b94d230e89212c4..52b441db86411ec826d3bcea33d0795a39eb2753 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -26,6 +26,7 @@
 #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
@@ -1265,6 +1266,8 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
   rtc::Md5Digest payload_checksum_;
 };
 
+class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {};
+
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
 TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480));
@@ -1489,9 +1492,47 @@ TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms_voip) {
       50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
+static const SdpAudioFormat kOpusFormat("opus", 48000, 2, {{"stereo", "1"}});

[ossu, 2017/03/14 20:25:11]

Replicate the tests for Opus, but using an SdpAudioFormat, so that we ensure the
output is consistent. Should I perhaps add similar versions of the other format
tests? Opus is the one with the most number of parameters that can go wrong, but
the others will eventually need changing over to the new API.

+
+TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms) {
+  std::unique_ptr<AudioEncoder> encoder(
+      new AudioEncoderOpus(120, kOpusFormat));
+  ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(encoder.get(), 120));
+  Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "855041f2490b887302bce9d544731849",
+          "855041f2490b887302bce9d544731849",
+          "9692eede45638eb425e0daf9c75b5c7a",
+          "86d3552bb3492247f965cdd0e88a1c82"),
+      AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "d781cce1ab986b618d0da87226cdde30",
+          "d781cce1ab986b618d0da87226cdde30",
+          "8d6782b905c3230d4b0e3e83e1fc3439",
+          "798347a685fac7d0c2d8f748ffe66881"),
+      50, test::AcmReceiveTestOldApi::kStereoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) {
+  std::unique_ptr<AudioEncoder> encoder(
+      new AudioEncoderOpus(120, kOpusFormat));
+  ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(encoder.get(), 120));
+  // If not set, default will be kAudio in case of stereo.
+  EXPECT_EQ(0, send_test_->acm()->SetOpusApplication(kVoip));
+  Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "9b9e12bc3cc793740966e11cbfa8b35b",
+          "9b9e12bc3cc793740966e11cbfa8b35b",
+          "0de6249018fdd316c21086db84e10610",
+          "9c4cb69db77b85841a5f8225bb8f508b"),
+      AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "c7340b1189652ab6b5e80dade7390cb4",
+          "c7340b1189652ab6b5e80dade7390cb4",
+          "95612864c954ee63e28cc6eebad56626",
+          "ae33ea2e43407cf9ebdabbbd6ca912a3"),
+      50, test::AcmReceiveTestOldApi::kStereoOutput);
+}
+
 // This test is for verifying the SetBitRate function. The bitrate is changed at
 // the beginning, and the number of generated bytes are checked.
-class AcmSetBitRateOldApi : public ::testing::Test {
+class AcmSetBitRateTest : public ::testing::Test {
  protected:
   static const int kTestDurationMs = 1000;
 
@@ -1521,11 +1562,12 @@ class AcmSetBitRateOldApi : public ::testing::Test {
                                      payload_type, frame_size_samples);
   }
 
-  // Runs the test. SetUpSender() and RegisterSendCodec() must have been called
-  // before calling this method.
-  void Run(int target_bitrate_bps, int expected_total_bits) {
-    ASSERT_TRUE(send_test_->acm());
-    send_test_->acm()->SetBitRate(target_bitrate_bps);
+    bool RegisterExternalSendCodec(AudioEncoder* external_speech_encoder,
+                                 int payload_type) {
+    return send_test_->RegisterExternalCodec(external_speech_encoder);
+  }
+
+  void RunInner(int expected_total_bits) {
     int nr_bytes = 0;
     while (std::unique_ptr<test::Packet> next_packet =
                send_test_->NextPacket()) {
@@ -1550,6 +1592,26 @@ class AcmSetBitRateOldApi : public ::testing::Test {
   std::unique_ptr<test::InputAudioFile> audio_source_;
 };
 
+class AcmSetBitRateOldApi : public AcmSetBitRateTest {
+ protected:
+  // Runs the test. SetUpSender() must have been called and a codec must be set
+  // up before calling this method.
+  void Run(int target_bitrate_bps, int expected_total_bits) {
+    ASSERT_TRUE(send_test_->acm());
+    send_test_->acm()->SetBitRate(target_bitrate_bps);
+    RunInner(expected_total_bits);
+  }
+};
+
+class AcmSetBitRateNewApi : public AcmSetBitRateTest {
+ protected:
+  // Runs the test. SetUpSender() must have been called and a codec must be set
+  // up before calling this method.
+  void Run(int expected_total_bits) {
+    RunInner(expected_total_bits);
+  }
+};
+
 TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
 #if defined(WEBRTC_ANDROID)
@@ -1568,18 +1630,56 @@ TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
 #endif  // WEBRTC_ANDROID
 }
 
+
 // The result on the Android platforms is inconsistent for this test case.
 // On android_rel the result is different from android and android arm64 rel.
 #if defined(WEBRTC_ANDROID)
 #define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
+#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
+  DISABLED_OpusFromFormat_48khz_20ms_100kbps
 #else
 #define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
+#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
+  OpusFromFormat_48khz_20ms_100kbps
 #endif
 TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
   Run(100000, 100888);
 }
 
+TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
+  std::unique_ptr<AudioEncoder> encoder(new AudioEncoderOpus(
+      107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}})));
+  ASSERT_TRUE(SetUpSender());
+  ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
+#if defined(WEBRTC_ANDROID)
+  RunInner(9288);
+#else
+  RunInner(9024);
+#endif  // WEBRTC_ANDROID
+}
+
+TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
+  std::unique_ptr<AudioEncoder> encoder(new AudioEncoderOpus(
+      107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}})));
+  ASSERT_TRUE(SetUpSender());
+  ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
+#if defined(WEBRTC_ANDROID)
+  RunInner(47960);
+#else
+  RunInner(49544);
+#endif  // WEBRTC_ANDROID
+}
+
+TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
+  std::unique_ptr<AudioEncoder> encoder(new AudioEncoderOpus(
+      107,
+      SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "100000"}})));
+  ASSERT_TRUE(SetUpSender());
+  ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
+  RunInner(100888);
+}
+
 // These next 2 tests ensure that the SetBitRate function has no effect on PCM
 TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_8kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));