Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc

Index: webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
index dce5f4c516ae91bd723614a4050ac7ef2ffe98c4..b9b0bd8fe9533290e3c879abf45e4145aeac6910 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -44,6 +44,17 @@ int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
   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.