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 7515903bcb1e63e0cd8698b8e6b41d5627c5f272..98569a6aeb887db3d5f1ed9d50c5735fb89c69cd 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"
@@ -1266,6 +1267,8 @@ 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));
@@ -1460,8 +1463,10 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
 
 #if WEBRTC_OPUS_SUPPORT_120MS_PTIME
 #define MAYBE_Opus_stereo_20ms DISABLED_Opus_stereo_20ms
+#define MAYBE_OpusFromFormat_stereo_20ms DISABLED_OpusFromFormat_stereo_20ms
 #else
 #define MAYBE_Opus_stereo_20ms Opus_stereo_20ms
+#define MAYBE_OpusFromFormat_stereo_20ms OpusFromFormat_stereo_20ms
 #endif
 TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
@@ -1478,10 +1483,30 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms) {
       50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
+TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms) {
+  const SdpAudioFormat kOpusFormat("opus", 48000, 2, {{"stereo", "1"}});
+  AudioEncoderOpus encoder(120, kOpusFormat);
+  ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(&encoder, 120));
+  Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "855041f2490b887302bce9d544731849",
+          "855041f2490b887302bce9d544731849",
+          "9692eede45638eb425e0daf9c75b5c7a",
+          "86d3552bb3492247f965cdd0e88a1c82"),
+      AcmReceiverBitExactnessOldApi::PlatformChecksum(
+          "d781cce1ab986b618d0da87226cdde30",
+          "d781cce1ab986b618d0da87226cdde30",
+          "8d6782b905c3230d4b0e3e83e1fc3439",
+          "798347a685fac7d0c2d8f748ffe66881"),
+      50, test::AcmReceiveTestOldApi::kStereoOutput);
+}
+
 #if WEBRTC_OPUS_SUPPORT_120MS_PTIME
 #define MAYBE_Opus_stereo_20ms_voip DISABLED_Opus_stereo_20ms_voip
+#define MAYBE_OpusFromFormat_stereo_20ms_voip \
+  DISABLED_OpusFromFormat_stereo_20ms_voip
 #else
 #define MAYBE_Opus_stereo_20ms_voip Opus_stereo_20ms_voip
+#define MAYBE_OpusFromFormat_stereo_20ms_voip OpusFromFormat_stereo_20ms_voip
 #endif
 TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms_voip) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
@@ -1500,9 +1525,28 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_Opus_stereo_20ms_voip) {
       50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
+TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms_voip) {
+  const SdpAudioFormat kOpusFormat("opus", 48000, 2, {{"stereo", "1"}});
+  AudioEncoderOpus encoder(120, kOpusFormat);
+  ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(&encoder, 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;
 
@@ -1532,11 +1576,12 @@ 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()) {
@@ -1561,10 +1606,31 @@ 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); }
+};
+
 #if WEBRTC_OPUS_SUPPORT_120MS_PTIME
 #define MAYBE_Opus_48khz_20ms_10kbps DISABLED_Opus_48khz_20ms_10kbps
+#define MAYBE_OpusFromFormat_48khz_20ms_10kbps \
+  DISABLED_OpusFromFormat_48khz_20ms_10kbps
 #else
 #define MAYBE_Opus_48khz_20ms_10kbps Opus_48khz_20ms_10kbps
+#define MAYBE_OpusFromFormat_48khz_20ms_10kbps OpusFromFormat_48khz_20ms_10kbps
 #endif
 TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
@@ -1575,10 +1641,25 @@ TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_10kbps) {
 #endif  // WEBRTC_ANDROID
 }
 
+TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_10kbps) {
+  AudioEncoderOpus encoder(
+      107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}}));
+  ASSERT_TRUE(SetUpSender());
+  ASSERT_TRUE(RegisterExternalSendCodec(&encoder, 107));
+#if defined(WEBRTC_ANDROID)
+  RunInner(9288);
+#else
+  RunInner(9024);
+#endif  // WEBRTC_ANDROID
+}
+
 #if WEBRTC_OPUS_SUPPORT_120MS_PTIME
 #define MAYBE_Opus_48khz_20ms_50kbps DISABLED_Opus_48khz_20ms_50kbps
+#define MAYBE_OpusFromFormat_48khz_20ms_50kbps \
+  DISABLED_OpusFromFormat_48khz_20ms_50kbps
 #else
 #define MAYBE_Opus_48khz_20ms_50kbps Opus_48khz_20ms_50kbps
+#define MAYBE_OpusFromFormat_48khz_20ms_50kbps OpusFromFormat_48khz_20ms_50kbps
 #endif
 TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
@@ -1589,18 +1670,42 @@ TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_50kbps) {
 #endif  // WEBRTC_ANDROID
 }
 
+TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_50kbps) {
+  AudioEncoderOpus encoder(
+      107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}}));
+  ASSERT_TRUE(SetUpSender());
+  ASSERT_TRUE(RegisterExternalSendCodec(&encoder, 107));
+#if defined(WEBRTC_ANDROID)
+  RunInner(47960);
+#else
+  RunInner(49544);
+#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) || WEBRTC_OPUS_SUPPORT_120MS_PTIME
 #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, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
+  AudioEncoderOpus encoder(
+      107, SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "100000"}}));
+  ASSERT_TRUE(SetUpSender());
+  ASSERT_TRUE(RegisterExternalSendCodec(&encoder, 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));