Add new fast mode for NetEq's Accelerate operation

This change instroduces a mode where the Accelerate operation will be
more aggressive. When enabled, it will allow acceleration at lower
correlation levels, and possibly remove multiple pitch periods at
once.

The feature is enabled through NetEq::Config, and is off by
default. This means that bit-exactness tests are currently not
affected.

A unit test was added for the Accelerate class, with and without fast
mode enabled.

BUG=4691
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50039004

Cr-Commit-Position: refs/heads/master@{#9295}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index fe078fd..1351e66 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -92,6 +92,7 @@
       decoder_error_code_(0),
       background_noise_mode_(config.background_noise_mode),
       playout_mode_(config.playout_mode),
+      enable_fast_accelerate_(config.enable_fast_accelerate),
       decoded_packet_sequence_number_(-1),
       decoded_packet_timestamp_(0) {
   LOG(LS_INFO) << "NetEq config: " << config.ToString();
@@ -745,9 +746,12 @@
       return_value = DoExpand(play_dtmf);
       break;
     }
-    case kAccelerate: {
+    case kAccelerate:
+    case kFastAccelerate: {
+      const bool fast_accelerate =
+          enable_fast_accelerate_ && (operation == kFastAccelerate);
       return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type,
-                                  play_dtmf);
+                                  play_dtmf, fast_accelerate);
       break;
     }
     case kPreemptiveExpand: {
@@ -956,9 +960,8 @@
   // Check if we already have enough samples in the |sync_buffer_|. If so,
   // change decision to normal, unless the decision was merge, accelerate, or
   // preemptive expand.
-  if (samples_left >= output_size_samples_ &&
-      *operation != kMerge &&
-      *operation != kAccelerate &&
+  if (samples_left >= output_size_samples_ && *operation != kMerge &&
+      *operation != kAccelerate && *operation != kFastAccelerate &&
       *operation != kPreemptiveExpand) {
     *operation = kNormal;
     return 0;
@@ -1034,8 +1037,9 @@
       decision_logic_->set_generated_noise_samples(0);
       return 0;
     }
-    case kAccelerate: {
-      // In order to do a accelerate we need at least 30 ms of audio data.
+    case kAccelerate:
+    case kFastAccelerate: {
+      // In order to do an accelerate we need at least 30 ms of audio data.
       if (samples_left >= samples_30_ms) {
         // Already have enough data, so we do not need to extract any more.
         decision_logic_->set_sample_memory(samples_left);
@@ -1124,13 +1128,13 @@
     }
   }
 
-  if (*operation == kAccelerate ||
+  if (*operation == kAccelerate || *operation == kFastAccelerate ||
       *operation == kPreemptiveExpand) {
     decision_logic_->set_sample_memory(samples_left + extracted_samples);
     decision_logic_->set_prev_time_scale(true);
   }
 
-  if (*operation == kAccelerate) {
+  if (*operation == kAccelerate || *operation == kFastAccelerate) {
     // Check that we have enough data (30ms) to do accelerate.
     if (extracted_samples + samples_left < samples_30_ms) {
       // TODO(hlundin): Write test for this.
@@ -1263,7 +1267,8 @@
     assert(sync_buffer_->Channels() == decoder->Channels());
     assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->Channels());
     assert(*operation == kNormal || *operation == kAccelerate ||
-           *operation == kMerge || *operation == kPreemptiveExpand);
+           *operation == kFastAccelerate || *operation == kMerge ||
+           *operation == kPreemptiveExpand);
     packet_list->pop_front();
     size_t payload_length = packet->payload_length;
     int16_t decode_length;
@@ -1427,9 +1432,11 @@
   return 0;
 }
 
-int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, size_t decoded_length,
+int NetEqImpl::DoAccelerate(int16_t* decoded_buffer,
+                            size_t decoded_length,
                             AudioDecoder::SpeechType speech_type,
-                            bool play_dtmf) {
+                            bool play_dtmf,
+                            bool fast_accelerate) {
   const size_t required_samples = 240 * fs_mult_;  // Must have 30 ms.
   size_t borrowed_samples_per_channel = 0;
   size_t num_channels = algorithm_buffer_->Channels();
@@ -1447,9 +1454,9 @@
   }
 
   int16_t samples_removed;
-  Accelerate::ReturnCodes return_code = accelerate_->Process(
-      decoded_buffer, decoded_length, algorithm_buffer_.get(),
-      &samples_removed);
+  Accelerate::ReturnCodes return_code =
+      accelerate_->Process(decoded_buffer, decoded_length, fast_accelerate,
+                           algorithm_buffer_.get(), &samples_removed);
   stats_.AcceleratedSamples(samples_removed);
   switch (return_code) {
     case Accelerate::kSuccess: