AcmReceiver: Look up last decoder in NetEq's table of decoders

webrtc/modules/audio_coding/acm2/acm_receiver.h

 /*
  *  Copyright (c) 2013 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 21 matching lines @@
 
 namespace webrtc {
 
 struct CodecInst;
 class NetEq;
 
 namespace acm2 {
 
 class AcmReceiver {
  public:
-  struct Decoder {
-    int acm_codec_id;
-    uint8_t payload_type;
-    // This field is meaningful for codecs where both mono and
-    // stereo versions are registered under the same ID.
-    size_t channels;
-    int sample_rate_hz;
-  };
-
   // Constructor of the class
   explicit AcmReceiver(const AudioCodingModule::Config& config);
 
   // Destructor of the class.
   ~AcmReceiver();
 
   //
   // Inserts a payload with its associated RTP-header into NetEq.
   //
   // Input:
@@ skipping 194 matching lines @@
   //    -round_trip_time_ms : estimate of the round-trip-time (in milliseconds).
   // Return value           : list of packets to be retransmitted.
   //
   std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
 
   //
   // Get statistics of calls to GetAudio().
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
  private:
-  const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
-                                    uint8_t payload_type) const
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+  struct Decoder {
+    int acm_codec_id;
+    uint8_t payload_type;
+    // This field is meaningful for codecs where both mono and
+    // stereo versions are registered under the same ID.
+    size_t channels;
+    int sample_rate_hz;
+  };
+
+  const rtc::Optional<CodecInst> RtpHeaderToDecoder(
+      const RTPHeader& rtp_header,
+      uint8_t first_payload_byte) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
   rtc::CriticalSection crit_sect_;
-  const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
+  rtc::Optional<CodecInst> last_audio_decoder_ GUARDED_BY(crit_sect_);
   ACMResampler resampler_ GUARDED_BY(crit_sect_);
   std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
   NetEq* neteq_;
   // Decoders map is keyed by payload type
   std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_);
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
   rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_