Move functionality out from AudioFrame and into AudioFrameOperations.

webrtc/modules/include/module_common_types.h

 /*
  *  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_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
 #define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
 
 #include <assert.h>
 #include <string.h>  // memcpy
 
 #include <algorithm>
 #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"
 
 namespace webrtc {
 
 struct RTPAudioHeader {
   uint8_t numEnergy;                  // number of valid entries in arrOfEnergy
   uint8_t arrOfEnergy[kRtpCsrcSize];  // one energy byte (0-9) per channel
   bool isCNG;                         // is this CNG
@@ skipping 482 matching lines @@
 /* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
  * allows for adding and subtracting frames while keeping track of the resulting
  * states.
  *
  * Notes
  * - The total number of samples in |data_| is
  *   samples_per_channel_ * num_channels_
  *
  * - 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:
   // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
   enum : size_t {
     kMaxDataSizeSamples = 3840
   };
 
   enum VADActivity {
     kVadActive = 0,
@@ skipping 14 matching lines @@
   // contents of |data_|).
   void Reset();
 
   void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
                    size_t samples_per_channel, int sample_rate_hz,
                    SpeechType speech_type, VADActivity vad_activity,
                    size_t num_channels = 1);
 
   void CopyFrom(const AudioFrame& src);
 
-  void Mute();
-
-  AudioFrame& operator>>=(const int rhs);
-  AudioFrame& operator+=(const AudioFrame& rhs);
+  // These methods are deprecated. Use the functions in
+  // webrtc/audio/utility instead. These methods will exists for a
+  // short period of time until webrtc clients have updated. See
+  // 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_;
-  int sample_rate_hz_;
-  size_t num_channels_;
-  SpeechType speech_type_;
-  VADActivity vad_activity_;
+  size_t samples_per_channel_ = 0;
+  int sample_rate_hz_ = 0;
+  size_t num_channels_ = 0;
+  SpeechType speech_type_ = kUndefined;
+  VADActivity vad_activity_ = kVadUnknown;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
 };
 
 // TODO(henrik.lundin) Can we remove the call to data_()?
 // See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
 inline AudioFrame::AudioFrame()
     : data_() {
-  Reset();
 }
 
 inline void AudioFrame::Reset() {
   id_ = -1;
   // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
   // to an invalid value, or add a new member to indicate invalidity.
   timestamp_ = 0;
   elapsed_time_ms_ = -1;
   ntp_time_ms_ = -1;
   samples_per_channel_ = 0;
@@ skipping 53 matching lines @@
 inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
   assert((num_channels_ > 0) && (num_channels_ < 3));
   if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
 
   for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
     data_[i] = static_cast<int16_t>(data_[i] >> 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));
   if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
   if (num_channels_ != rhs.num_channels_) return *this;
 
   bool noPrevData = false;
   if (samples_per_channel_ != rhs.samples_per_channel_) {
     if (samples_per_channel_ == 0) {
       // special case we have no data to start with
@@ skipping 13 matching lines @@
   if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
 
   if (noPrevData) {
     memcpy(data_, rhs.data_,
            sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
   } else {
     // IMPROVEMENT this can be done very fast in assembly
     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;
 }
 
 inline bool IsNewerSequenceNumber(uint16_t sequence_number,
                                   uint16_t prev_sequence_number) {
   // Distinguish between elements that are exactly 0x8000 apart.
   // If s1>s2 and |s1-s2| = 0x8000: IsNewer(s1,s2)=true, IsNewer(s2,s1)=false
   // rather than having IsNewer(s1,s2) = IsNewer(s2,s1) = false.
@@ skipping 63 matching lines @@
     return unwrapped;
   }
 
  private:
   int64_t last_seq_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_