Move functionality out from AudioFrame and into AudioFrameOperations.

This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.

Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.

The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.

TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}

.gn

@@ -25,6 +25,7 @@ check_targets = [
   "//webrtc/api:audio_mixer_api",
   "//webrtc/api:transport_api",
   "//webrtc/api:rtc_stats_api",
+  "//webrtc/audio/utility/*",
   "//webrtc/modules/audio_coding:audio_decoder_factory_interface",
   "//webrtc/modules/audio_coding:audio_format",
   "//webrtc/modules/audio_coding:audio_format_conversion",

webrtc/audio/BUILD.gn

@@ -37,11 +37,6 @@ rtc_static_library("audio") {
     "../modules/audio_processing",
     "../system_wrappers",
     "../voice_engine",
-    "utility",  # Bogus dep, needed for landing
-                # codereview.webrtc.org/2424173003 without breaking
-                # internal projects. See
-                # bugs.webrtc.org/6548. TODO(aleloi): remove dependency
-                # when codereview.webrtc.org/2424173003 has landed.
   ]
 }
 if (rtc_include_tests) {
@@ -51,6 +46,7 @@ if (rtc_include_tests) {
       "audio_receive_stream_unittest.cc",
       "audio_send_stream_unittest.cc",
       "audio_state_unittest.cc",
+      "utility/audio_frame_operations_unittest.cc",
     ]
     deps = [
       ":audio",
@@ -59,6 +55,8 @@ if (rtc_include_tests) {
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer:audio_mixer_impl",
       "../test:test_common",
+      "../test:test_support",
+      "utility:audio_frame_operations",
       "//testing/gmock",
       "//testing/gtest",
     ]

webrtc/audio/utility/BUILD.gn

@@ -8,4 +8,20 @@
 import("../../build/webrtc.gni")
 
 group("utility") {
+  public_deps = [
+    ":audio_frame_operations",
+  ]
+}
+
+rtc_static_library("audio_frame_operations") {
+  sources = [
+    "audio_frame_operations.cc",
+    "audio_frame_operations.h",
+  ]
+
+  deps = [
+    "../..:webrtc_common",
+    "../../base:rtc_base_approved",
+    "../../modules/audio_coding:audio_format_conversion",
+  ]
 }

webrtc/audio/utility/audio_frame_operations.cc

@@ -8,9 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/audio/utility/audio_frame_operations.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
+
+#include <algorithm>
+
 #include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 namespace {
@@ -19,7 +23,50 @@ namespace  {
 const size_t kMuteFadeFrames = 128;
 const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
 
-}  // namespace {
+}  // namespace
+
+void AudioFrameOperations::Add(const AudioFrame& frame_to_add,
+                               AudioFrame* result_frame) {
+  // Sanity check.
+  RTC_DCHECK(result_frame);
+  RTC_DCHECK_GT(result_frame->num_channels_, 0);
+  RTC_DCHECK_EQ(result_frame->num_channels_, frame_to_add.num_channels_);
+
+  bool no_previous_data = false;
+  if (result_frame->samples_per_channel_ != frame_to_add.samples_per_channel_) {
+    // Special case we have no data to start with.
+    RTC_DCHECK_EQ(result_frame->samples_per_channel_, 0);
+    result_frame->samples_per_channel_ = frame_to_add.samples_per_channel_;
+    no_previous_data = true;
+  }
+
+  if (result_frame->vad_activity_ == AudioFrame::kVadActive ||
+      frame_to_add.vad_activity_ == AudioFrame::kVadActive) {
+    result_frame->vad_activity_ = AudioFrame::kVadActive;
+  } else if (result_frame->vad_activity_ == AudioFrame::kVadUnknown ||
+             frame_to_add.vad_activity_ == AudioFrame::kVadUnknown) {
+    result_frame->vad_activity_ = AudioFrame::kVadUnknown;
+  }
+
+  if (result_frame->speech_type_ != frame_to_add.speech_type_)
+    result_frame->speech_type_ = AudioFrame::kUndefined;
+
+  if (no_previous_data) {
+    std::copy(frame_to_add.data_, frame_to_add.data_ +
+                                      frame_to_add.samples_per_channel_ *
+                                          result_frame->num_channels_,
+              result_frame->data_);
+  } else {
+    for (size_t i = 0;
+         i < result_frame->samples_per_channel_ * result_frame->num_channels_;
+         i++) {
+      const int32_t wrap_guard = static_cast<int32_t>(result_frame->data_[i]) +
+                           static_cast<int32_t>(frame_to_add.data_[i]);
+      result_frame->data_[i] = rtc::saturated_cast<int16_t>(wrap_guard);
+    }
+  }
+  return;
+}
 
 void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
                                         size_t samples_per_channel,
@@ -68,7 +115,10 @@ int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
 }
 
 void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
-  if (frame->num_channels_ != 2) return;
+  RTC_DCHECK(frame);
+  if (frame->num_channels_ != 2) {
+    return;
+  }
 
   for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
     int16_t temp_data = frame->data_[i];
@@ -77,7 +127,8 @@ void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
   }
 }
 
-void AudioFrameOperations::Mute(AudioFrame* frame, bool previous_frame_muted,
+void AudioFrameOperations::Mute(AudioFrame* frame,
+                                bool previous_frame_muted,
                                 bool current_frame_muted) {
   RTC_DCHECK(frame);
   if (!previous_frame_muted && !current_frame_muted) {
@@ -125,14 +176,30 @@ void AudioFrameOperations::Mute(AudioFrame* frame, bool previous_frame_muted,
   }
 }
 
+void AudioFrameOperations::Mute(AudioFrame* frame) {
+  Mute(frame, true, true);
+}
+
+void AudioFrameOperations::ApplyHalfGain(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK_GT(frame->num_channels_, 0);
+  if (frame->num_channels_ < 1) {
+    return;
+  }
+
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       i++) {
+    frame->data_[i] = frame->data_[i] >> 1;
+  }
+}
+
 int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
   if (frame.num_channels_ != 2) {
     return -1;
   }
 
   for (size_t i = 0; i < frame.samples_per_channel_; i++) {
-    frame.data_[2 * i] =
+    frame.data_[2 * i] = static_cast<int16_t>(left * frame.data_[2 * i]);
-        static_cast<int16_t>(left * frame.data_[2 * i]);
     frame.data_[2 * i + 1] =
         static_cast<int16_t>(right * frame.data_[2 * i + 1]);
   }
@@ -156,5 +223,4 @@ int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) {
   }
   return 0;
 }
-
 }  // namespace webrtc

webrtc/audio/utility/audio_frame_operations.h

@@ -0,0 +1,83 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
+#define WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
+
+#include <stddef.h>
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioFrame;
+
+// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
+// Change reference parameters to pointers. Consider using a namespace rather
+// than a class.
+class AudioFrameOperations {
+ public:
+  // Add samples in |frame_to_add| with samples in |result_frame|
+  // putting the results in |results_frame|.  The fields
+  // |vad_activity_| and |speech_type_| of the result frame are
+  // updated. If |result_frame| is empty (|samples_per_channel_|==0),
+  // the samples in |frame_to_add| are added to it.  The number of
+  // channels and number of samples per channel must match except when
+  // |result_frame| is empty.
+  static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
+
+  // Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
+  // operation, meaning src_audio and dst_audio must point to different
+  // buffers. It is the caller's responsibility to ensure that |dst_audio| is
+  // sufficiently large.
+  static void MonoToStereo(const int16_t* src_audio,
+                           size_t samples_per_channel,
+                           int16_t* dst_audio);
+  // |frame.num_channels_| will be updated. This version checks for sufficient
+  // buffer size and that |num_channels_| is mono.
+  static int MonoToStereo(AudioFrame* frame);
+
+  // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
+  // operation, meaning |src_audio| and |dst_audio| may point to the same
+  // buffer.
+  static void StereoToMono(const int16_t* src_audio,
+                           size_t samples_per_channel,
+                           int16_t* dst_audio);
+  // |frame.num_channels_| will be updated. This version checks that
+  // |num_channels_| is stereo.
+  static int StereoToMono(AudioFrame* frame);
+
+  // Swap the left and right channels of |frame|. Fails silently if |frame| is
+  // not stereo.
+  static void SwapStereoChannels(AudioFrame* frame);
+
+  // Conditionally zero out contents of |frame| for implementing audio mute:
+  //  |previous_frame_muted| &&  |current_frame_muted| - Zero out whole frame.
+  //  |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
+  // !|previous_frame_muted| &&  |current_frame_muted| - Fade-out at frame end.
+  // !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
+  static void Mute(AudioFrame* frame,
+                   bool previous_frame_muted,
+                   bool current_frame_muted);
+
+  // Zero out contents of frame.
+  static void Mute(AudioFrame* frame);
+
+  // Halve samples in |frame|.
+  static void ApplyHalfGain(AudioFrame* frame);
+
+  static int Scale(float left, float right, AudioFrame& frame);
+
+  static int ScaleWithSat(float scale, AudioFrame& frame);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_

webrtc/audio/utility/audio_frame_operations_unittest.cc

@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
@@ -365,5 +365,25 @@ TEST_F(AudioFrameOperationsTest, ScaleWithSatSucceeds) {
   VerifyFramesAreEqual(scaled_frame, frame_);
 }
 
+TEST_F(AudioFrameOperationsTest, AddingXToEmptyGivesX) {
+  // When samples_per_channel_ is 0, the frame counts as empty and zero.
+  AudioFrame frame_to_add_to;
+  frame_to_add_to.samples_per_channel_ = 0;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  VerifyFramesAreEqual(frame_, frame_to_add_to);
+}
+
+TEST_F(AudioFrameOperationsTest, AddingTwoFramesProducesTheirSum) {
+  AudioFrame frame_to_add_to;
+  frame_to_add_to.samples_per_channel_ = frame_.samples_per_channel_;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+  SetFrameData(&frame_to_add_to, 1000);
+
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  SetFrameData(&frame_, frame_.data_[0] + 1000);
+  VerifyFramesAreEqual(frame_, frame_to_add_to);
+}
 }  // namespace
 }  // namespace webrtc

webrtc/modules/BUILD.gn

@@ -483,7 +483,6 @@ if (rtc_include_tests) {
       "rtp_rtcp/test/testAPI/test_api_audio.cc",
       "rtp_rtcp/test/testAPI/test_api_rtcp.cc",
       "rtp_rtcp/test/testAPI/test_api_video.cc",
-      "utility/source/audio_frame_operations_unittest.cc",
       "utility/source/file_player_unittests.cc",
       "utility/source/process_thread_impl_unittest.cc",
       "video_coding/codecs/test/packet_manipulator_unittest.cc",

webrtc/modules/audio_conference_mixer/BUILD.gn

@@ -39,8 +39,8 @@ rtc_static_library("audio_conference_mixer") {
   }
 
   deps = [
+    "../../audio/utility:audio_frame_operations",
     "../../system_wrappers",
     "../audio_processing",
-    "../utility",
   ]
 }

webrtc/modules/audio_conference_mixer/DEPS

@@ -1,4 +1,5 @@
 include_rules = [
+  "+webrtc/audio/utility/audio_frame_operations.h",
   "+webrtc/base",
   "+webrtc/system_wrappers",
 ]

webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
@@ -38,9 +38,9 @@ typedef std::list<ParticipantFrameStruct*> ParticipantFrameStructList;
 void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
   assert(mixed_frame->num_channels_ >= frame->num_channels_);
   if (use_limiter) {
-    // Divide by two to avoid saturation in the mixing.
+    // This is to avoid saturation in the mixing. It is only
-    // This is only meaningful if the limiter will be used.
+    // meaningful if the limiter will be used.
-    *frame >>= 1;
+    AudioFrameOperations::ApplyHalfGain(frame);
   }
   if (mixed_frame->num_channels_ > frame->num_channels_) {
     // We only support mono-to-stereo.
@@ -49,7 +49,7 @@ void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
     AudioFrameOperations::MonoToStereo(frame);
   }
 
-  *mixed_frame += *frame;
+  AudioFrameOperations::Add(*frame, mixed_frame);
 }
 
 // Return the max number of channels from a |list| composed of AudioFrames.
@@ -303,7 +303,7 @@ void AudioConferenceMixerImpl::Process() {
         if(mixedAudio->samples_per_channel_ == 0) {
             // Nothing was mixed, set the audio samples to silence.
             mixedAudio->samples_per_channel_ = _sampleSize;
-            mixedAudio->Mute();
+            AudioFrameOperations::Mute(mixedAudio);
         } else {
             // Only call the limiter if we have something to mix.
             LimitMixedAudio(mixedAudio);
@@ -922,7 +922,7 @@ bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame* mixedAudio) const {
     //
     // Instead we double the frame (with addition since left-shifting a
     // negative value is undefined).
-    *mixedAudio += *mixedAudio;
+    AudioFrameOperations::Add(*mixedAudio, mixedAudio);
 
     if(error != _limiter->kNoError) {
         WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,

webrtc/modules/audio_mixer/BUILD.gn

@@ -32,10 +32,10 @@ rtc_static_library("audio_mixer_impl") {
   deps = [
     ":audio_frame_manipulator",
     "../..:webrtc_common",
+    "../../audio/utility:audio_frame_operations",
     "../../base:rtc_base_approved",
-    "../../modules/audio_processing",
-    "../../modules/utility",
     "../../system_wrappers",
+    "../audio_processing",
   ]
 }
 
@@ -51,7 +51,7 @@ rtc_static_library("audio_frame_manipulator") {
   ]
 
   deps = [
+    "../../audio/utility",
     "../../base:rtc_base_approved",
-    "../../modules/utility",
   ]
 }

webrtc/modules/audio_mixer/DEPS

@@ -1,4 +1,5 @@
 include_rules = [
+  "+webrtc/audio/utility/audio_frame_operations.h",
   "+webrtc/base",
   "+webrtc/call",
   "+webrtc/common_audio",

webrtc/modules/audio_mixer/audio_frame_manipulator.cc

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
 
 namespace webrtc {
 

webrtc/modules/audio_mixer/audio_mixer_impl.cc

@@ -14,9 +14,9 @@
 #include <functional>
 #include <utility>
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
 
 namespace webrtc {
 namespace {
@@ -106,12 +106,12 @@ int32_t MixFromList(AudioFrame* mixed_audio,
     // Mix |f.frame| into |mixed_audio|, with saturation protection.
     // These effect is applied to |f.frame| itself prior to mixing.
     if (use_limiter) {
-      // Divide by two to avoid saturation in the mixing.
+      // This is to avoid saturation in the mixing. It is only
-      // This is only meaningful if the limiter will be used.
+      // meaningful if the limiter will be used.
-      *frame >>= 1;
+      AudioFrameOperations::ApplyHalfGain(frame);
     }
     RTC_DCHECK_EQ(frame->num_channels_, mixed_audio->num_channels_);
-    *mixed_audio += *frame;
+    AudioFrameOperations::Add(*frame, mixed_audio);
   }
   return 0;
 }
@@ -250,7 +250,7 @@ void AudioMixerImpl::Mix(size_t number_of_channels,
   if (audio_frame_for_mixing->samples_per_channel_ == 0) {
     // Nothing was mixed, set the audio samples to silence.
     audio_frame_for_mixing->samples_per_channel_ = sample_size_;
-    audio_frame_for_mixing->Mute();
+    AudioFrameOperations::Mute(audio_frame_for_mixing);
   } else {
     // Only call the limiter if we have something to mix.
     LimitMixedAudio(audio_frame_for_mixing);
@@ -357,7 +357,7 @@ bool AudioMixerImpl::LimitMixedAudio(AudioFrame* mixed_audio) const {
   //
   // Instead we double the frame (with addition since left-shifting a
   // negative value is undefined).
-  *mixed_audio += *mixed_audio;
+  AudioFrameOperations::Add(*mixed_audio, mixed_audio);
 
   if (error != limiter_->kNoError) {
     LOG_F(LS_ERROR) << "Error from AudioProcessing: " << error;

webrtc/modules/audio_processing/BUILD.gn

@@ -165,6 +165,7 @@ rtc_static_library("audio_processing") {
   defines = []
   deps = [
     "../..:webrtc_common",
+    "../../audio/utility:audio_frame_operations",
     "../audio_coding:isac",
   ]
 

webrtc/modules/audio_processing/DEPS

@@ -1,4 +1,5 @@
 include_rules = [
+  "+webrtc/audio/utility/audio_frame_operations.h",
   "+webrtc/base",
   "+webrtc/common_audio",
   "+webrtc/system_wrappers",

webrtc/modules/audio_processing/vad/standalone_vad.cc

@@ -10,9 +10,9 @@
 
 #include "webrtc/modules/audio_processing/vad/standalone_vad.h"
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {

webrtc/modules/include/module_common_types.h

@@ -18,6 +18,8 @@
 #include <limits>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/deprecation.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/rotation.h"
 #include "webrtc/typedefs.h"
@@ -520,8 +522,6 @@ class CallStatsObserver {
  *
  * - Stereo data is interleaved starting with the left channel.
  *
- * - The +operator assume that you would never add exactly opposite frames when
- *   deciding the resulting state. To do this use the -operator.
  */
 class AudioFrame {
  public:
@@ -556,26 +556,29 @@ class AudioFrame {
 
   void CopyFrom(const AudioFrame& src);
 
-  void Mute();
+  // These methods are deprecated. Use the functions in
-
+  // webrtc/audio/utility instead. These methods will exists for a
-  AudioFrame& operator>>=(const int rhs);
+  // short period of time until webrtc clients have updated. See
-  AudioFrame& operator+=(const AudioFrame& rhs);
+  // webrtc:6548 for details.
+  RTC_DEPRECATED void Mute();
+  RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
+  RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);
 
   int id_;
   // RTP timestamp of the first sample in the AudioFrame.
-  uint32_t timestamp_;
+  uint32_t timestamp_ = 0;
   // Time since the first frame in milliseconds.
   // -1 represents an uninitialized value.
-  int64_t elapsed_time_ms_;
+  int64_t elapsed_time_ms_ = -1;
   // NTP time of the estimated capture time in local timebase in milliseconds.
   // -1 represents an uninitialized value.
-  int64_t ntp_time_ms_;
+  int64_t ntp_time_ms_ = -1;
   int16_t data_[kMaxDataSizeSamples];
-  size_t samples_per_channel_;
+  size_t samples_per_channel_ = 0;
-  int sample_rate_hz_;
+  int sample_rate_hz_ = 0;
-  size_t num_channels_;
+  size_t num_channels_ = 0;
-  SpeechType speech_type_;
+  SpeechType speech_type_ = kUndefined;
-  VADActivity vad_activity_;
+  VADActivity vad_activity_ = kVadUnknown;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
@@ -585,7 +588,6 @@ class AudioFrame {
 // See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
 inline AudioFrame::AudioFrame()
     : data_() {
-  Reset();
 }
 
 inline void AudioFrame::Reset() {
@@ -659,18 +661,6 @@ inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
   return *this;
 }
 
-namespace {
-inline int16_t ClampToInt16(int32_t input) {
-  if (input < -0x00008000) {
-    return -0x8000;
-  } else if (input > 0x00007FFF) {
-    return 0x7FFF;
-  } else {
-    return static_cast<int16_t>(input);
-  }
-}
-}
-
 inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
   // Sanity check
   assert((num_channels_ > 0) && (num_channels_ < 3));
@@ -704,7 +694,7 @@ inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
     for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
       int32_t wrap_guard =
           static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
-      data_[i] = ClampToInt16(wrap_guard);
+      data_[i] = rtc::saturated_cast<int16_t>(wrap_guard);
     }
   }
   return *this;

webrtc/modules/module_common_types_unittest.cc

@@ -112,15 +112,6 @@ TEST(LatestTimestamp, Wrap) {
   EXPECT_EQ(0x0000FFFFu, LatestTimestamp(0xFFFF0000, 0x0000FFFF));
 }
 
-TEST(ClampToInt16, TestCases) {
-  EXPECT_EQ(0x0000, ClampToInt16(0x00000000));
-  EXPECT_EQ(0x0001, ClampToInt16(0x00000001));
-  EXPECT_EQ(0x7FFF, ClampToInt16(0x00007FFF));
-  EXPECT_EQ(0x7FFF, ClampToInt16(0x7FFFFFFF));
-  EXPECT_EQ(-0x0001, ClampToInt16(-0x00000001));
-  EXPECT_EQ(-0x8000, ClampToInt16(-0x8000));
-  EXPECT_EQ(-0x8000, ClampToInt16(-0x7FFFFFFF));
-}
 
 TEST(SequenceNumberUnwrapper, Limits) {
   SequenceNumberUnwrapper unwrapper;

webrtc/modules/utility/BUILD.gn

@@ -16,7 +16,6 @@ rtc_static_library("utility") {
     "include/helpers_android.h",
     "include/jvm_android.h",
     "include/process_thread.h",
-    "source/audio_frame_operations.cc",
     "source/coder.cc",
     "source/coder.h",
     "source/file_player.cc",
@@ -38,10 +37,13 @@ rtc_static_library("utility") {
 
   deps = [
     "../..:webrtc_common",
+    "../../audio/utility:audio_frame_operations",
     "../../base:rtc_task_queue",
     "../../common_audio",
     "../../system_wrappers",
     "../audio_coding",
+    "../audio_coding:builtin_audio_decoder_factory",
+    "../audio_coding:rent_a_codec",
     "../media_file",
   ]
 }

webrtc/modules/utility/DEPS

@@ -1,4 +1,7 @@
 include_rules = [
+  # TODO(aleloi): remove when clients update. See
+  # bugs.webrtc.org/6548.
+  "+webrtc/audio/utility/audio_frame_operations.h",
   "+webrtc/base",
   "+webrtc/common_audio",
   "+webrtc/common_video",

webrtc/modules/utility/include/audio_frame_operations.h

@@ -10,54 +10,11 @@
 
 #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
 #define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+// The contents of this file have moved to
+// //webrtc/audio/utility. This file is deprecated.
 
-#include "webrtc/typedefs.h"
+// TODO(aleloi): Remove this file when clients have updated their
-
+// includes. See bugs.webrtc.org/6548.
-namespace webrtc {
+#include "webrtc/audio/utility/audio_frame_operations.h"
-
-class AudioFrame;
-
-// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
-// Change reference parameters to pointers. Consider using a namespace rather
-// than a class.
-class AudioFrameOperations {
- public:
-  // Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
-  // operation, meaning src_audio and dst_audio must point to different
-  // buffers. It is the caller's responsibility to ensure that |dst_audio| is
-  // sufficiently large.
-  static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel,
-                           int16_t* dst_audio);
-  // |frame.num_channels_| will be updated. This version checks for sufficient
-  // buffer size and that |num_channels_| is mono.
-  static int MonoToStereo(AudioFrame* frame);
-
-  // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
-  // operation, meaning |src_audio| and |dst_audio| may point to the same
-  // buffer.
-  static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel,
-                           int16_t* dst_audio);
-  // |frame.num_channels_| will be updated. This version checks that
-  // |num_channels_| is stereo.
-  static int StereoToMono(AudioFrame* frame);
-
-  // Swap the left and right channels of |frame|. Fails silently if |frame| is
-  // not stereo.
-  static void SwapStereoChannels(AudioFrame* frame);
-
-  // Conditionally zero out contents of |frame| for implementing audio mute:
-  //  |previous_frame_muted| &&  |current_frame_muted| - Zero out whole frame.
-  //  |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
-  // !|previous_frame_muted| &&  |current_frame_muted| - Fade-out at frame end.
-  // !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
-  static void Mute(AudioFrame* frame, bool previous_frame_muted,
-                   bool current_frame_muted);
-
-  static int Scale(float left, float right, AudioFrame& frame);
-
-  static int ScaleWithSat(float scale, AudioFrame& frame);
-};
-
-}  // namespace webrtc
 
 #endif  // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_

webrtc/voice_engine/BUILD.gn

@@ -90,6 +90,7 @@ rtc_static_library("voice_engine") {
     "../api:audio_mixer_api",
     "../api:call_api",
     "../api:transport_api",
+    "../audio/utility:audio_frame_operations",
     "../base:rtc_base_approved",
     "../common_audio",
     "../logging:rtc_event_log_api",

webrtc/voice_engine/DEPS

@@ -1,4 +1,5 @@
 include_rules = [
+  "+webrtc/audio/utility/audio_frame_operations.h",
   "+webrtc/base",
   "+webrtc/call",
   "+webrtc/common_audio",

webrtc/voice_engine/channel.cc

@@ -13,6 +13,7 @@
 #include <algorithm>
 #include <utility>
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/criticalsection.h"
@@ -32,7 +33,6 @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
@@ -613,7 +613,7 @@ MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
     // TODO(henrik.lundin): We should be able to do better than this. But we
     // will have to go through all the cases below where the audio samples may
     // be used, and handle the muted case in some way.
-    audioFrame->Mute();
+    AudioFrameOperations::Mute(audioFrame);
   }
 
   // Convert module ID to internal VoE channel ID

webrtc/voice_engine/output_mixer.cc

@@ -10,9 +10,9 @@
 
 #include "webrtc/voice_engine/output_mixer.h"
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"

webrtc/voice_engine/transmit_mixer.cc

@@ -12,9 +12,9 @@
 
 #include <memory>
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"

webrtc/voice_engine/utility.cc

@@ -10,13 +10,13 @@
 
 #include "webrtc/voice_engine/utility.h"
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {

webrtc/voice_engine/voe_external_media_impl.cc

@@ -10,6 +10,7 @@
 
 #include "webrtc/voice_engine/voe_external_media_impl.h"
 
+#include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
@@ -149,7 +150,7 @@ int VoEExternalMediaImpl::GetAudioFrame(int channel, int desired_sample_rate_hz,
       desired_sample_rate_hz == 0 ? -1 : desired_sample_rate_hz;
   auto ret = channelPtr->GetAudioFrameWithMuted(channel, frame);
   if (ret == MixerParticipant::AudioFrameInfo::kMuted) {
-    frame->Mute();
+    AudioFrameOperations::Mute(frame);
   }
   return ret == MixerParticipant::AudioFrameInfo::kError ? -1 : 0;
 }