Injectable audio encoders: BuiltinAudioEncoderFactory

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc

 /*
  *  Copyright (c) 2014 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 <stdio.h>
 #include <string.h>
 #include <memory>
 #include <vector>
 
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/md5digest.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_coding/acm2/acm_receive_test.h"
 #include "webrtc/modules/audio_coding/acm2/acm_send_test.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #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"
 #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
@@ skipping 1219 matching lines @@
   std::unique_ptr<test::AcmSendTestOldApi> send_test_;
   std::unique_ptr<test::InputAudioFile> audio_source_;
   uint32_t frame_size_rtp_timestamps_;
   int packet_count_;
   uint8_t payload_type_;
   uint16_t last_sequence_number_;
   uint32_t last_timestamp_;
   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));
   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "0b58f9eeee43d5891f5f6c75e77984a3",
           "c7e5bdadfa2871df95639fcc297cf23d",
           "0499ca260390769b3172136faad925b9",
           "866abf524acd2807efbe65e133c23f95"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "3c79f16f34218271f3dca4e2b1dfe1bb",
@@ skipping 204 matching lines @@
           "0de6249018fdd316c21086db84e10610",
           "9c4cb69db77b85841a5f8225bb8f508b"),
       AcmReceiverBitExactnessOldApi::PlatformChecksum(
           "c7340b1189652ab6b5e80dade7390cb4",
           "c7340b1189652ab6b5e80dade7390cb4",
           "95612864c954ee63e28cc6eebad56626",
           "ae33ea2e43407cf9ebdabbbd6ca912a3"),
       50, test::AcmReceiveTestOldApi::kStereoOutput);
 }
 
+static const SdpAudioFormat kOpusFormat("opus", 48000, 2, {{"stereo", "1"}});
+
+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;
 
   // Sets up the test::AcmSendTest object. Returns true on success, otherwise
   // false.
   bool SetUpSender() {
     const std::string input_file_name =
         webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
     // Note that |audio_source_| will loop forever. The test duration is set
     // explicitly by |kTestDurationMs|.
     audio_source_.reset(new test::InputAudioFile(input_file_name));
     static const int kSourceRateHz = 32000;
     send_test_.reset(new test::AcmSendTestOldApi(
         audio_source_.get(), kSourceRateHz, kTestDurationMs));
     return send_test_.get();
   }
 
   // Registers a send codec in the test::AcmSendTest object. Returns true on
   // success, false on failure.
   virtual bool RegisterSendCodec(const char* payload_name,
                                  int sampling_freq_hz,
                                  int channels,
                                  int payload_type,
                                  int frame_size_samples,
                                  int frame_size_rtp_timestamps) {
     return send_test_->RegisterCodec(payload_name, sampling_freq_hz, channels,
                                      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) {

[kwiberg-webrtc, 2017/02/19 21:41:10]

clang-format

[ossu, 2017/02/20 12:20:25]

Alright.

+    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()) {
       nr_bytes += next_packet->payload_length_bytes();
     }
     EXPECT_EQ(expected_total_bits, nr_bytes * 8);
   }
 
   void SetUpTest(const char* codec_name,
                  int codec_sample_rate_hz,
                  int channels,
                  int payload_type,
                  int codec_frame_size_samples,
                  int codec_frame_size_rtp_timestamps) {
     ASSERT_TRUE(SetUpSender());
     ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels,
                                   payload_type, codec_frame_size_samples,
                                   codec_frame_size_rtp_timestamps));
   }
 
   std::unique_ptr<test::AcmSendTestOldApi> send_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)
   Run(10000, 9288);
 #else
   Run(10000, 9024);
 #endif  // WEBRTC_ANDROID
 }
 
 TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
 #if defined(WEBRTC_ANDROID)
   Run(50000, 47960);
 #else
   Run(50000, 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)
 #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));
   Run(8000, 128000);
 }
 
 TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
   Run(32000, 128000);
 }
@@ skipping 246 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc