Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.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 26 matching lines @@
                                        size_t encoded_len,
                                        int sample_rate_hz,
                                        int16_t* decoded,
                                        SpeechType* speech_type) {
   RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
   size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
   *speech_type = ConvertSpeechType(1);
   return static_cast<int>(ret);
 }
 
+AudioDecoder::PacketSplits AudioDecoderPcm16B::SplitPacket(
+    const uint8_t* payload,
+    size_t payload_len) const {
+  // TODO(ossu): Investigate if we can ever get 44.1KHz audio here, in which
+  //             case rounding will break. Consider replacing with
+  //             CheckedDivExact to catch that happening.

[kwiberg-webrtc, 2016/08/26 12:39:25]

Could you handle that by letting SplitBySamples work on 10ms instead of 1ms
chunks?

+  return SplitBySamples(payload, payload_len,
+                        sample_rate_hz_ * 2 * num_channels_ / 1000,
+                        sample_rate_hz_ / 1000);
+}
+
 int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
                                        size_t encoded_len) const {
   // Two encoded byte per sample per channel.
   return static_cast<int>(encoded_len / (2 * Channels()));
 }
 
 }  // namespace webrtc