1 files changed, 25 insertions, 80 deletions

diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 3a5a13ff..c95214b2 100644
--- a/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -26,7 +26,7 @@
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "webrtc/system_wrappers/interface/data_log.h"
@@ -140,17 +140,20 @@ class AudioDecoderTest : public ::testing::Test {
                           size_t input_len_samples,
                           uint8_t* output) {
     size_t enc_len_bytes = 0;
-    for (int i = 0; i < audio_encoder_->num_10ms_frames_per_packet(); ++i) {
+    scoped_ptr<int16_t[]> interleaved_input(
+        new int16_t[channels_ * input_len_samples]);
+    for (int i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
       EXPECT_EQ(0u, enc_len_bytes);
-      EXPECT_TRUE(audio_encoder_->Encode(0,
-                                         input,
-                                         audio_encoder_->sample_rate_hz() / 100,
-                                         data_length_ * 2,
-                                         output,
-                                         &enc_len_bytes,
-                                         &output_timestamp_));
+
+      // Duplicate the mono input signal to however many channels the test
+      // wants.
+      test::InputAudioFile::DuplicateInterleaved(
+          input, input_len_samples, channels_, interleaved_input.get());
+
+      EXPECT_TRUE(audio_encoder_->Encode(
+          0, interleaved_input.get(), audio_encoder_->sample_rate_hz() / 100,
+          data_length_ * 2, output, &enc_len_bytes, &output_timestamp_));
     }
-    EXPECT_EQ(input_len_samples, enc_len_bytes);
     return static_cast<int>(enc_len_bytes);
   }
 
@@ -295,7 +298,7 @@ class AudioDecoderPcm16BTest : public AudioDecoderTest {
     codec_input_rate_hz_ = 8000;
     frame_size_ = 160;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderPcm16B(kDecoderPCM16B);
+    decoder_ = new AudioDecoderPcm16B;
     assert(decoder_);
   }
 
@@ -366,7 +369,7 @@ class AudioDecoderIsacFloatTest : public AudioDecoderTest {
     input_size_ = 160;
     frame_size_ = 480;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderIsac;
+    decoder_ = new AudioDecoderIsac(16000);
     assert(decoder_);
     WebRtcIsac_Create(&encoder_);
     WebRtcIsac_SetEncSampRate(encoder_, 16000);
@@ -404,7 +407,7 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest {
     input_size_ = 320;
     frame_size_ = 960;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderIsacSwb;
+    decoder_ = new AudioDecoderIsac(32000);
     assert(decoder_);
     WebRtcIsac_Create(&encoder_);
     WebRtcIsac_SetEncSampRate(encoder_, 32000);
@@ -435,19 +438,6 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest {
   int input_size_;
 };
 
-// This test is identical to AudioDecoderIsacSwbTest, except that it creates
-// an AudioDecoderIsacFb decoder object.
-class AudioDecoderIsacFbTest : public AudioDecoderIsacSwbTest {
- protected:
-  AudioDecoderIsacFbTest() : AudioDecoderIsacSwbTest() {
-    // Delete the |decoder_| that was created by AudioDecoderIsacSwbTest and
-    // create an AudioDecoderIsacFb object instead.
-    delete decoder_;
-    decoder_ = new AudioDecoderIsacFb;
-    assert(decoder_);
-  }
-};
-
 class AudioDecoderIsacFixTest : public AudioDecoderTest {
  protected:
   AudioDecoderIsacFixTest() : AudioDecoderTest() {
@@ -635,57 +625,23 @@ class AudioDecoderOpusTest : public AudioDecoderTest {
     codec_input_rate_hz_ = 48000;
     frame_size_ = 480;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderOpus(kDecoderOpus);
-    assert(decoder_);
-    WebRtcOpus_EncoderCreate(&encoder_, 1);
+    decoder_ = new AudioDecoderOpus(1);
+    AudioEncoderOpus::Config config;
+    config.frame_size_ms = static_cast<int>(frame_size_) / 48;
+    audio_encoder_.reset(new AudioEncoderOpus(config));
   }
-
-  ~AudioDecoderOpusTest() {
-    WebRtcOpus_EncoderFree(encoder_);
-  }
-
-  virtual void InitEncoder() {}
-
-  virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
-                          uint8_t* output) OVERRIDE {
-    int enc_len_bytes = WebRtcOpus_Encode(encoder_, const_cast<int16_t*>(input),
-        static_cast<int16_t>(input_len_samples),
-        static_cast<int16_t>(data_length_), output);
-    EXPECT_GT(enc_len_bytes, 0);
-    return enc_len_bytes;
-  }
-
-  OpusEncInst* encoder_;
 };
 
 class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
  protected:
   AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() {
     channels_ = 2;
-    WebRtcOpus_EncoderFree(encoder_);
     delete decoder_;
-    decoder_ = new AudioDecoderOpus(kDecoderOpus_2ch);
-    assert(decoder_);
-    WebRtcOpus_EncoderCreate(&encoder_, 2);
-  }
-
-  virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
-                          uint8_t* output) OVERRIDE {
-    // Create stereo by duplicating each sample in |input|.
-    const int input_stereo_samples = static_cast<int>(input_len_samples) * 2;
-    scoped_ptr<int16_t[]> input_stereo(new int16_t[input_stereo_samples]);
-    test::InputAudioFile::DuplicateInterleaved(
-        input, input_len_samples, 2, input_stereo.get());
-
-    // Note that the input length is given as samples per channel.
-    int enc_len_bytes =
-        WebRtcOpus_Encode(encoder_,
-                          input_stereo.get(),
-                          static_cast<int16_t>(input_len_samples),
-                          static_cast<int16_t>(data_length_),
-                          output);
-    EXPECT_GT(enc_len_bytes, 0);
-    return enc_len_bytes;
+    decoder_ = new AudioDecoderOpus(2);
+    AudioEncoderOpus::Config config;
+    config.frame_size_ms = static_cast<int>(frame_size_) / 48;
+    config.num_channels = 2;
+    audio_encoder_.reset(new AudioEncoderOpus(config));
   }
 };
 
@@ -752,17 +708,6 @@ TEST_F(AudioDecoderIsacSwbTest, EncodeDecode) {
   DecodePlcTest();
 }
 
-TEST_F(AudioDecoderIsacFbTest, EncodeDecode) {
-  int tolerance = 19757;
-  double mse = 8.18e6;
-  int delay = 160;  // Delay from input to output.
-  EXPECT_TRUE(AudioDecoder::CodecSupported(kDecoderISACswb));
-  EncodeDecodeTest(0, tolerance, mse, delay);
-  ReInitTest();
-  EXPECT_TRUE(decoder_->HasDecodePlc());
-  DecodePlcTest();
-}
-
 TEST_F(AudioDecoderIsacFixTest, DISABLED_EncodeDecode) {
   int tolerance = 11034;
   double mse = 3.46e6;