Convert CNG into C++ and remove it from AudioDecoder

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc

Index: webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 3b48131a754e7f33ce8ded0d5cb259e584d7ae93..c7b0e303eb9cbce3b191129853de8abb8a73950c 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -21,19 +21,6 @@ namespace {
 
 const int kMaxFrameSizeMs = 60;
 
-std::unique_ptr<CNG_enc_inst, CngInstDeleter> CreateCngInst(
-    int sample_rate_hz,
-    int sid_frame_interval_ms,
-    int num_cng_coefficients) {
-  CNG_enc_inst* ci;
-  RTC_CHECK_EQ(0, WebRtcCng_CreateEnc(&ci));
-  std::unique_ptr<CNG_enc_inst, CngInstDeleter> cng_inst(ci);
-  RTC_CHECK_EQ(0,
-               WebRtcCng_InitEnc(cng_inst.get(), sample_rate_hz,
-                                 sid_frame_interval_ms, num_cng_coefficients));
-  return cng_inst;
-}
-
 }  // namespace
 
 AudioEncoderCng::Config::Config() = default;
@@ -75,9 +62,10 @@ AudioEncoderCng::AudioEncoderCng(Config&& config)
       sid_frame_interval_ms_(config.sid_frame_interval_ms),
       last_frame_active_(true),
       vad_(config.vad ? std::unique_ptr<Vad>(config.vad)
-                      : CreateVad(config.vad_mode)) {
-  cng_inst_ = CreateCngInst(SampleRateHz(), sid_frame_interval_ms_,
-                            num_cng_coefficients_);
+           : CreateVad(config.vad_mode)),
+      cng_encoder_(new ComfortNoiseEncoder(SampleRateHz(),
+                                           sid_frame_interval_ms_,
+                                           num_cng_coefficients_)) {
 }
 
 AudioEncoderCng::~AudioEncoderCng() = default;
@@ -187,8 +175,9 @@ void AudioEncoderCng::Reset() {
   rtp_timestamps_.clear();
   last_frame_active_ = true;
   vad_->Reset();
-  cng_inst_ = CreateCngInst(SampleRateHz(), sid_frame_interval_ms_,
-                            num_cng_coefficients_);
+  cng_encoder_.reset(
+      new ComfortNoiseEncoder(SampleRateHz(), sid_frame_interval_ms_,
+                              num_cng_coefficients_));

[kwiberg-webrtc, 2016/04/12 13:35:30]

A small utility function in an anonymous namespace that returns a
unique_ptr<ComfortNoiseEncoder>?

[ossu, 2016/04/12 13:54:49]

Actually, there's a Reset method in it right now, that I bet I could use... not
sure it should remain in, though.

 }
 
 bool AudioEncoderCng::SetFec(bool enable) {
@@ -230,13 +219,13 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
         // WebRtcCng_Encode(), since later loop iterations may return zero in
         // that value, in which case we don't want to overwrite any value from
         // an earlier iteration.
-        size_t encoded_bytes_tmp = 0;
-        RTC_CHECK_GE(
-            WebRtcCng_Encode(cng_inst_.get(),
-                             &speech_buffer_[i * samples_per_10ms_frame],
-                             samples_per_10ms_frame, encoded.data(),
-                             &encoded_bytes_tmp, force_sid),
-            0);
+        int encoded_bytes_tmp =
+            cng_encoder_->Encode(
+                rtc::ArrayView<const int16_t>(
+                    &speech_buffer_[i * samples_per_10ms_frame],
+                    samples_per_10ms_frame),
+                encoded, force_sid);
+        RTC_CHECK_GE(encoded_bytes_tmp, 0);

[kwiberg-webrtc, 2016/04/12 13:35:30]

Use rtc::Optional<size_t> if you need to return an error. But as I said, I agree
that Encode can probably just explode if called with invalid inputs.

Also, consider passing the Buffer* into Encode instead of an ArrayView. That
way, you completely remove the possibility of it being too small.

[ossu, 2016/04/12 13:54:49]

Yeah, Buffer is probably better. It's what we want to move to in general,
internally, anyway, right?

[hlundin-webrtc, 2016/04/13 07:05:23]

I'm fine with using a Buffer as well. But bear in mind that CNG is a special
breed. The length of the encoded data has no correlation with the number of
samples inserted. It will just estimate a predetermined number of noise
parameters from the input.

[kwiberg-webrtc, 2016/04/14 09:42:47]

Meaning its output length is a small constant? It probably doesn't hurt to use a
Buffer anyway---that way, callers won't depend on that constant, and the
interface is simpler.

         if (encoded_bytes_tmp > 0) {
           RTC_CHECK(!output_produced);
           info.encoded_bytes = encoded_bytes_tmp;