Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.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.
  */
 
 #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;
 }
 
 size_t AudioDecoderPcmU::Channels() const {
   return num_channels_;
 }
 
 int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
   RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }
 
 int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
                                      size_t encoded_len) const {
   // One encoded byte per sample per channel.
   return static_cast<int>(encoded_len / Channels());
 }
 
 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;
 }
 
 size_t AudioDecoderPcmA::Channels() const {
   return num_channels_;
 }
 
 int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
                                      size_t encoded_len,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
   RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }
 
 int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
                                      size_t encoded_len) const {
   // One encoded byte per sample per channel.
   return static_cast<int>(encoded_len / Channels());
 }
 
 }  // namespace webrtc