AudioDecoder: New method SampleRateHz, + implementations for our codecs

webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc

 /*
  *  Copyright (c) 2015 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 17 matching lines @@
 
 bool AudioDecoderG722::HasDecodePlc() const {
   return false;
 }
 
 int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  RTC_DCHECK_EQ(sample_rate_hz, 16000);
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret =
       WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }
 
 void AudioDecoderG722::Reset() {
   WebRtcG722_DecoderInit(dec_state_);
 }
 
 int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
                                      size_t encoded_len) const {
   // 1/2 encoded byte per sample per channel.
   return static_cast<int>(2 * encoded_len / Channels());
 }
 
+int AudioDecoderG722::SampleRateHz() const {
+  return 16000;
+}
+
 size_t AudioDecoderG722::Channels() const {
   return 1;
 }
 
 AudioDecoderG722Stereo::AudioDecoderG722Stereo() {
   WebRtcG722_CreateDecoder(&dec_state_left_);
   WebRtcG722_CreateDecoder(&dec_state_right_);
   WebRtcG722_DecoderInit(dec_state_left_);
   WebRtcG722_DecoderInit(dec_state_right_);
 }
 
 AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
   WebRtcG722_FreeDecoder(dec_state_left_);
   WebRtcG722_FreeDecoder(dec_state_right_);
 }
 
 int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
                                            size_t encoded_len,
                                            int sample_rate_hz,
                                            int16_t* decoded,
                                            SpeechType* speech_type) {
-  RTC_DCHECK_EQ(sample_rate_hz, 16000);
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   // De-interleave the bit-stream into two separate payloads.
   uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
   SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
   // Decode left and right.
   size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
                                          encoded_len / 2, decoded, &temp_type);
   size_t ret = WebRtcG722_Decode(
       dec_state_right_, &encoded_deinterleaved[encoded_len / 2],
       encoded_len / 2, &decoded[decoded_len], &temp_type);
   if (ret == decoded_len) {
     ret += decoded_len;  // Return total number of samples.
     // Interleave output.
     for (size_t k = ret / 2; k < ret; k++) {
       int16_t temp = decoded[k];
       memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
               (ret - k - 1) * sizeof(int16_t));
       decoded[2 * k - ret + 1] = temp;
     }
   }
   *speech_type = ConvertSpeechType(temp_type);
   delete[] encoded_deinterleaved;
   return static_cast<int>(ret);
 }
 
+int AudioDecoderG722Stereo::SampleRateHz() const {
+  return 16000;
+}
+
 size_t AudioDecoderG722Stereo::Channels() const {
   return 2;
 }
 
 void AudioDecoderG722Stereo::Reset() {
   WebRtcG722_DecoderInit(dec_state_left_);
   WebRtcG722_DecoderInit(dec_state_right_);
 }
 
 // Split the stereo packet and place left and right channel after each other
@@ skipping 16 matching lines @@
   // where N is the total number of samples.
   for (size_t i = 0; i < encoded_len / 2; i++) {
     uint8_t right_byte = encoded_deinterleaved[i + 1];
     memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
             encoded_len - i - 2);
     encoded_deinterleaved[encoded_len - 1] = right_byte;
   }
 }
 
 }  // namespace webrtc