Revert of Remove various IDs

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.
  */
 
@@ skipping 312 matching lines @@
   AudioFrame();
 
   // Resets all members to their default state.
   void Reset();
   // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
   // the buffer to be zeroed on the next call to mutable_data(). Callers
   // intending to write to the buffer immediately after Reset() can instead use
   // ResetWithoutMuting() to skip this wasteful zeroing.
   void ResetWithoutMuting();
 
-  // TODO(solenberg): Remove once downstream users of AudioFrame have updated.
-  RTC_DEPRECATED
-      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) {
-    RTC_UNUSED(id);
-    UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
-                speech_type, vad_activity, num_channels);
-  }
-
-  void UpdateFrame(uint32_t timestamp, const int16_t* data,
+  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);
 
   // Sets a wall-time clock timestamp in milliseconds to be used for profiling
   // of time between two points in the audio chain.
   // Example:
   //   t0: UpdateProfileTimeStamp()
@@ skipping 15 matching lines @@
   // Frame is muted by default.
   bool muted() const;
 
   // 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 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_ = 0;
   // Time since the first frame in milliseconds.
   // -1 represents an uninitialized value.
   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_ = -1;
   size_t samples_per_channel_ = 0;
   int sample_rate_hz_ = 0;
@@ skipping 27 matching lines @@
   // Visual Studio doesn't like this in the class definition.
   static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
 }
 
 inline void AudioFrame::Reset() {
   ResetWithoutMuting();
   muted_ = true;
 }
 
 inline void AudioFrame::ResetWithoutMuting() {
+  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;
   sample_rate_hz_ = 0;
   num_channels_ = 0;
   speech_type_ = kUndefined;
   vad_activity_ = kVadUnknown;
   profile_timestamp_ms_ = 0;
 }
 
-inline void AudioFrame::UpdateFrame(uint32_t timestamp,
+inline void AudioFrame::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) {
+  id_ = id;
   timestamp_ = timestamp;
   samples_per_channel_ = samples_per_channel;
   sample_rate_hz_ = sample_rate_hz;
   speech_type_ = speech_type;
   vad_activity_ = vad_activity;
   num_channels_ = num_channels;
 
   const size_t length = samples_per_channel * num_channels;
   assert(length <= kMaxDataSizeSamples);
   if (data != nullptr) {
     memcpy(data_, data, sizeof(int16_t) * length);
     muted_ = false;
   } else {
     muted_ = true;
   }
 }
 
 inline void AudioFrame::CopyFrom(const AudioFrame& src) {
   if (this == &src) return;
 
+  id_ = src.id_;
   timestamp_ = src.timestamp_;
   elapsed_time_ms_ = src.elapsed_time_ms_;
   ntp_time_ms_ = src.ntp_time_ms_;
   muted_ = src.muted();
   samples_per_channel_ = src.samples_per_channel_;
   sample_rate_hz_ = src.sample_rate_hz_;
   speech_type_ = src.speech_type_;
   vad_activity_ = src.vad_activity_;
   num_channels_ = src.num_channels_;
 
@@ skipping 198 matching lines @@
   static constexpr int kNotAProbe = -1;
   int send_bitrate_bps = -1;
   int probe_cluster_id = kNotAProbe;
   int probe_cluster_min_probes = -1;
   int probe_cluster_min_bytes = -1;
 };
 
 }  // namespace webrtc
 
 #endif  // MODULES_INCLUDE_MODULE_COMMON_TYPES_H_