Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc

Index: webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
index af164c4bb72b0b25525b3999f3e6887390a0b802..f2fdb1f15cdf94bb45b1b6ad2ac6f08ebd72f785 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -10,12 +10,21 @@
 
 #include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 
+#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 #include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 
 namespace webrtc {
 
 void AudioDecoderPcmU::Reset() {}
 
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp,
+    bool is_primary) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, is_primary, 8 * num_channels_, 8);
+}
+
 int AudioDecoderPcmU::SampleRateHz() const {
   return 8000;
 }
@@ -44,6 +53,14 @@ int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
 
 void AudioDecoderPcmA::Reset() {}
 
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp,
+    bool is_primary) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, is_primary, 8 * num_channels_, 8);
+}
+
 int AudioDecoderPcmA::SampleRateHz() const {
   return 8000;
 }