Revert of Prevent Opus DTX from generating intermittent noise during silence

webrtc/modules/audio_coding/codecs/opus/opus_interface.c

 /*
  *  Copyright (c) 2012 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 "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 
-#include <assert.h>
 #include <stdlib.h>
 #include <string.h>
 
 enum {
   /* Maximum supported frame size in WebRTC is 60 ms. */
   kWebRtcOpusMaxEncodeFrameSizeMs = 60,
 
   /* The format allows up to 120 ms frames. Since we don't control the other
    * side, we must allow for packets of that size. NetEq is currently limited
    * to 60 ms on the receive side. */
   kWebRtcOpusMaxDecodeFrameSizeMs = 120,
 
   /* Maximum sample count per channel is 48 kHz * maximum frame size in
    * milliseconds. */
   kWebRtcOpusMaxFrameSizePerChannel = 48 * kWebRtcOpusMaxDecodeFrameSizeMs,
 
   /* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */
   kWebRtcOpusDefaultFrameSize = 960,
-
-  // Maximum number of consecutive zeros, beyond or equal to which DTX can fail.
-  kZeroBreakCount = 157,
-
-#if defined(OPUS_FIXED_POINT)
-  kZeroBreakValue = 10,
-#else
-  kZeroBreakValue = 1,
-#endif
 };
 
 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
                                  int32_t channels,
                                  int32_t application) {
-  int opus_app;
-  if (!inst)
-    return -1;
+  OpusEncInst* state;
+  if (inst != NULL) {
+    state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst));
+    if (state) {
+      int opus_app;
+      switch (application) {
+        case 0: {
+          opus_app = OPUS_APPLICATION_VOIP;
+          break;
+        }
+        case 1: {
+          opus_app = OPUS_APPLICATION_AUDIO;
+          break;
+        }
+        default: {
+          free(state);
+          return -1;
+        }
+      }
 
-  switch (application) {
-    case 0:
-      opus_app = OPUS_APPLICATION_VOIP;
-      break;
-    case 1:
-      opus_app = OPUS_APPLICATION_AUDIO;
-      break;
-    default:
-      return -1;
+      int error;
+      state->encoder = opus_encoder_create(48000, channels, opus_app,
+                                           &error);
+      state->in_dtx_mode = 0;
+      if (error == OPUS_OK && state->encoder != NULL) {
+        *inst = state;
+        return 0;
+      }
+      free(state);
+    }
   }
-
-  OpusEncInst* state = calloc(1, sizeof(OpusEncInst));
-  assert(state);
-
-  // Allocate zero counters.
-  state->zero_counts = calloc(channels, sizeof(size_t));
-  assert(state->zero_counts);
-
-  int error;
-  state->encoder = opus_encoder_create(48000, channels, opus_app,
-                                       &error);
-  if (error != OPUS_OK || !state->encoder) {
-    WebRtcOpus_EncoderFree(state);
-    return -1;
-  }
-
-  state->in_dtx_mode = 0;
-  state->channels = channels;
-
-  *inst = state;
-  return 0;
+  return -1;
 }
 
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
   if (inst) {
     opus_encoder_destroy(inst->encoder);
-    free(inst->zero_counts);
     free(inst);
     return 0;
   } else {
     return -1;
   }
 }
 
 int WebRtcOpus_Encode(OpusEncInst* inst,
                       const int16_t* audio_in,
                       size_t samples,
                       size_t length_encoded_buffer,
                       uint8_t* encoded) {
   int res;
-  size_t i;
-  int c;
-
-  int16_t buffer[2 * 48 * kWebRtcOpusMaxEncodeFrameSizeMs];
 
   if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
     return -1;
   }
 
-  const int channels = inst->channels;
-  int use_buffer = 0;
-
-  // Break long consecutive zeros by forcing a "1" every |kZeroBreakCount|
-  // samples.
-  if (inst->in_dtx_mode) {
-    for (i = 0; i < samples; ++i) {
-      for (c = 0; c < channels; ++c) {
-        if (audio_in[i * channels + c] == 0) {
-          ++inst->zero_counts[c];
-          if (inst->zero_counts[c] == kZeroBreakCount) {
-            if (!use_buffer) {
-              memcpy(buffer, audio_in, samples * channels * sizeof(int16_t));
-              use_buffer = 1;
-            }
-            buffer[i * channels + c] = kZeroBreakValue;
-            inst->zero_counts[c] = 0;
-          }
-        } else {
-          inst->zero_counts[c] = 0;
-        }
-      }
-    }
-  }
-
   res = opus_encode(inst->encoder,
-                    use_buffer ? buffer : audio_in,
+                    (const opus_int16*)audio_in,
                     (int)samples,
                     encoded,
                     (opus_int32)length_encoded_buffer);
 
   if (res == 1) {
     // Indicates DTX since the packet has nothing but a header. In principle,
     // there is no need to send this packet. However, we do transmit the first
     // occurrence to let the decoder know that the encoder enters DTX mode.
     if (inst->in_dtx_mode) {
       return 0;
@@ skipping 348 matching lines @@
     return 0;
   }
 
   for (n = 0; n < channels; n++) {
     if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1)))
       return 1;
   }
 
   return 0;
 }