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..e600d2d56c1ab5947e8c5ce02cb782086ab83c76 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -11,6 +11,7 @@
 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 
 namespace webrtc {
@@ -44,6 +45,16 @@ int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
   return static_cast<int>(ret);
 }
 
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp,
+    bool is_primary) {
+  const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz_, 1000);
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, is_primary,
+      samples_per_ms * 2 * num_channels_, samples_per_ms);
+}
+
 int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
                                        size_t encoded_len) const {
   // Two encoded byte per sample per channel.