Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.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/g722/audio_decoder_g722.h"
 
 #include <string.h>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
 AudioDecoderG722::AudioDecoderG722() {
   WebRtcG722_CreateDecoder(&dec_state_);
   WebRtcG722_DecoderInit(dec_state_);
 }
 
 AudioDecoderG722::~AudioDecoderG722() {
@@ skipping 14 matching lines @@
   size_t ret =
       WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return static_cast<int>(ret);
 }
 
 void AudioDecoderG722::Reset() {
   WebRtcG722_DecoderInit(dec_state_);
 }
 
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp,
+    bool is_primary) {
+  return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
+                                                 timestamp, is_primary, 8, 16);
+}
+
 int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
                                      size_t encoded_len) const {
   // 1/2 encoded byte per sample per channel.
   return static_cast<int>(2 * encoded_len / Channels());
 }
 
 int AudioDecoderG722::SampleRateHz() const {
   return 16000;
 }
 
@@ skipping 50 matching lines @@
 
 size_t AudioDecoderG722Stereo::Channels() const {
   return 2;
 }
 
 void AudioDecoderG722Stereo::Reset() {
   WebRtcG722_DecoderInit(dec_state_left_);
   WebRtcG722_DecoderInit(dec_state_right_);
 }
 
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722Stereo::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp,
+    bool is_primary) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, is_primary, 2 * 8, 16);
+}
+
 // Split the stereo packet and place left and right channel after each other
 // in the output array.
 void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
                                                size_t encoded_len,
                                                uint8_t* encoded_deinterleaved) {
   // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
   // where "lx" is 4 bits representing left sample number x, and "rx" right
   // sample. Two samples fit in one byte, represented with |...|.
   for (size_t i = 0; i + 1 < encoded_len; i += 2) {
     uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
     encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
     encoded_deinterleaved[i + 1] = right_byte;
   }
 
   // Move one byte representing right channel each loop, and place it at the
   // end of the bytestream vector. After looping the data is reordered to:
   // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
   // where N is the total number of samples.
   for (size_t i = 0; i < encoded_len / 2; i++) {
     uint8_t right_byte = encoded_deinterleaved[i + 1];
     memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
             encoded_len - i - 2);
     encoded_deinterleaved[encoded_len - 1] = right_byte;
   }
 }
 
 }  // namespace webrtc