Simple cleanups of AudioDecoder and AudioEncoder classes

  * Make sure they're all final and don't allow copying or assignment.

  * Get rid of the single-channel PCM decoder classes.

  * Move some includes from .h to .cc files where possible.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1353803002

Cr-Commit-Position: refs/heads/master@{#10021}

3 files changed, 26 insertions, 26 deletions

diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
index 52f327bc61..afad9b1d6b 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -183,13 +183,13 @@ AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type) {
   }
   switch (codec_type) {
     case kDecoderPCMu:
-      return new AudioDecoderPcmU;
+      return new AudioDecoderPcmU(1);
     case kDecoderPCMa:
-      return new AudioDecoderPcmA;
+      return new AudioDecoderPcmA(1);
     case kDecoderPCMu_2ch:
-      return new AudioDecoderPcmUMultiCh(2);
+      return new AudioDecoderPcmU(2);
     case kDecoderPCMa_2ch:
-      return new AudioDecoderPcmAMultiCh(2);
+      return new AudioDecoderPcmA(2);
 #ifdef WEBRTC_CODEC_ILBC
     case kDecoderILBC:
       return new AudioDecoderIlbc;
@@ -207,14 +207,14 @@ AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type) {
     case kDecoderPCM16Bwb:
     case kDecoderPCM16Bswb32kHz:
     case kDecoderPCM16Bswb48kHz:
-      return new AudioDecoderPcm16B;
+      return new AudioDecoderPcm16B(1);
     case kDecoderPCM16B_2ch:
     case kDecoderPCM16Bwb_2ch:
     case kDecoderPCM16Bswb32kHz_2ch:
     case kDecoderPCM16Bswb48kHz_2ch:
-      return new AudioDecoderPcm16BMultiCh(2);
+      return new AudioDecoderPcm16B(2);
     case kDecoderPCM16B_5ch:
-      return new AudioDecoderPcm16BMultiCh(5);
+      return new AudioDecoderPcm16B(5);
 #ifdef WEBRTC_CODEC_G722
     case kDecoderG722:
       return new AudioDecoderG722;
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 537f16cca4..78ebf9f708 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -286,7 +286,7 @@ class AudioDecoderPcmUTest : public AudioDecoderTest {
   AudioDecoderPcmUTest() : AudioDecoderTest() {
     frame_size_ = 160;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderPcmU;
+    decoder_ = new AudioDecoderPcmU(1);
     AudioEncoderPcmU::Config config;
     config.frame_size_ms = static_cast<int>(frame_size_ / 8);
     config.payload_type = payload_type_;
@@ -299,7 +299,7 @@ class AudioDecoderPcmATest : public AudioDecoderTest {
   AudioDecoderPcmATest() : AudioDecoderTest() {
     frame_size_ = 160;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderPcmA;
+    decoder_ = new AudioDecoderPcmA(1);
     AudioEncoderPcmA::Config config;
     config.frame_size_ms = static_cast<int>(frame_size_ / 8);
     config.payload_type = payload_type_;
@@ -313,7 +313,7 @@ class AudioDecoderPcm16BTest : public AudioDecoderTest {
     codec_input_rate_hz_ = 16000;
     frame_size_ = 20 * codec_input_rate_hz_ / 1000;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderPcm16B;
+    decoder_ = new AudioDecoderPcm16B(1);
     assert(decoder_);
     AudioEncoderPcm16B::Config config;
     config.sample_rate_hz = codec_input_rate_hz_;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index e3a0cc17af..e9ebe064eb 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -10,7 +10,6 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h"
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 
@@ -21,16 +20,14 @@ using ::testing::_;
 using ::testing::SetArgPointee;
 using ::testing::Return;
 
-
-class MockAudioDecoderOpus : public AudioDecoderOpus {
+class MockAudioDecoder final : public AudioDecoder {
  public:
   static const int kPacketDuration = 960;  // 48 kHz * 20 ms
 
-  explicit MockAudioDecoderOpus(int num_channels)
-      : AudioDecoderOpus(num_channels),
-        fec_enabled_(false) {
+  explicit MockAudioDecoder(size_t num_channels)
+      : num_channels_(num_channels), fec_enabled_(false) {
   }
-  virtual ~MockAudioDecoderOpus() { Die(); }
+  ~MockAudioDecoder() override { Die(); }
   MOCK_METHOD0(Die, void());
 
   MOCK_METHOD0(Reset, void());
@@ -49,6 +46,8 @@ class MockAudioDecoderOpus : public AudioDecoderOpus {
     return fec_enabled_;
   }
 
+  size_t Channels() const override { return num_channels_; }
+
   void set_fec_enabled(bool enable_fec) { fec_enabled_ = enable_fec; }
 
   bool fec_enabled() const { return fec_enabled_; }
@@ -75,13 +74,14 @@ class MockAudioDecoderOpus : public AudioDecoderOpus {
   }
 
  private:
+  const size_t num_channels_;
   bool fec_enabled_;
 };
 
 class NetEqNetworkStatsTest : public NetEqExternalDecoderTest {
  public:
   static const int kPayloadSizeByte = 30;
-  static const int kFrameSizeMs = 20;  // frame size of Opus
+  static const int kFrameSizeMs = 20;
   static const int kMaxOutputSize = 960;  // 10 ms * 48 kHz * 2 channels.
 
 enum logic {
@@ -108,7 +108,7 @@ struct NetEqNetworkStatsCheck {
 };
 
   NetEqNetworkStatsTest(NetEqDecoder codec,
-                        MockAudioDecoderOpus* decoder)
+                        MockAudioDecoder* decoder)
       : NetEqExternalDecoderTest(codec, decoder),
         external_decoder_(decoder),
         samples_per_ms_(CodecSampleRateHz(codec) / 1000),
@@ -227,7 +227,7 @@ struct NetEqNetworkStatsCheck {
     expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 1065;
     RunTest(50, expects);
 
-    // Next we enable Opus FEC.
+    // Next we enable FEC.
     external_decoder_->set_fec_enabled(true);
     // If FEC fills in the lost packets, no packet loss will be counted.
     expects.stats_ref.packet_loss_rate = 0;
@@ -261,7 +261,7 @@ struct NetEqNetworkStatsCheck {
   }
 
  private:
-  MockAudioDecoderOpus* external_decoder_;
+  MockAudioDecoder* external_decoder_;
   const int samples_per_ms_;
   const size_t frame_size_samples_;
   rtc::scoped_ptr<test::RtpGenerator> rtp_generator_;
@@ -272,22 +272,22 @@ struct NetEqNetworkStatsCheck {
   int16_t output_[kMaxOutputSize];
 };
 
-TEST(NetEqNetworkStatsTest, OpusDecodeFec) {
-  MockAudioDecoderOpus decoder(1);
+TEST(NetEqNetworkStatsTest, DecodeFec) {
+  MockAudioDecoder decoder(1);
   NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
-TEST(NetEqNetworkStatsTest, StereoOpusDecodeFec) {
-  MockAudioDecoderOpus decoder(2);
+TEST(NetEqNetworkStatsTest, StereoDecodeFec) {
+  MockAudioDecoder decoder(2);
   NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
   test.DecodeFecTest();
   EXPECT_CALL(decoder, Die()).Times(1);
 }
 
 TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
-  MockAudioDecoderOpus decoder(1);
+  MockAudioDecoder decoder(1);
   NetEqNetworkStatsTest test(kDecoderOpus, &decoder);
   test.NoiseExpansionTest();
   EXPECT_CALL(decoder, Die()).Times(1);