NetEq: Don't check sample rate and frame size upon error

webrtc/modules/audio_coding/neteq/include/neteq.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 147 matching lines @@
   // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
   // can be implied by inserting a sync-packet.
   // Returns kOk on success, kFail on failure.
   virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                                uint32_t receive_timestamp) = 0;
 
   // Instructs NetEq to deliver 10 ms of audio data. The data is written to
   // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |speech_type_|,
   // |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and
   // |vad_activity_| are updated upon success. If an error is returned, some
-  // fields may not have been updated.
+  // fields may not have been updated, or may contain inconsistent values.
   // If muted state is enabled (through Config::enable_muted_state), |muted|
   // may be set to true after a prolonged expand period. When this happens, the
   // |data_| in |audio_frame| is not written, but should be interpreted as being
   // all zeros.
   // Returns kOK on success, or kFail in case of an error.
   virtual int GetAudio(AudioFrame* audio_frame, bool* muted) = 0;
 
   // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
   // information in the codec database. Returns 0 on success, -1 on failure.
   // The name is only used to provide information back to the caller about the
@@ skipping 122 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_