Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.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/ilbc/audio_decoder_ilbc.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
 
 namespace webrtc {
 
 AudioDecoderIlbc::AudioDecoderIlbc() {
   WebRtcIlbcfix_DecoderCreate(&dec_state_);
   WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 
 AudioDecoderIlbc::~AudioDecoderIlbc() {
   WebRtcIlbcfix_DecoderFree(dec_state_);
@@ skipping 17 matching lines @@
 }
 
 size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) {
   return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
 }
 
 void AudioDecoderIlbc::Reset() {
   WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 
+std::vector<AudioDecoder::ParseResult> AudioDecoderIlbc::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp,
+    bool is_primary) {
+  std::vector<ParseResult> results;
+  size_t bytes_per_frame;
+  int timestamps_per_frame;
+  if (payload.size() >= 950) {
+    LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Payload too large";
+    return results;
+  }
+  if (payload.size() % 38 == 0) {
+    // 20 ms frames.
+    bytes_per_frame = 38;
+    timestamps_per_frame = 160;
+  } else if (payload.size() % 50 == 0) {
+    // 30 ms frames.
+    bytes_per_frame = 50;
+    timestamps_per_frame = 240;
+  } else {
+    LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Invalid payload";
+    return results;
+  }
+
+  RTC_DCHECK_EQ(0u, payload.size() % bytes_per_frame);
+  if (payload.size() == bytes_per_frame) {
+    std::unique_ptr<EncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(this, std::move(payload), is_primary));
+    results.emplace_back(timestamp, is_primary, std::move(frame));
+  } else {
+    size_t byte_offset;
+    uint32_t timestamp_offset;
+    for (byte_offset = 0, timestamp_offset = 0;
+         byte_offset < payload.size();
+         byte_offset += bytes_per_frame,
+             timestamp_offset += timestamps_per_frame) {
+      rtc::Buffer new_payload(payload.data() + byte_offset, bytes_per_frame);
+      std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
+          this, std::move(new_payload), is_primary));
+      results.emplace_back(timestamp + timestamp_offset, is_primary,
+                           std::move(frame));
+    }
+  }
+
+  return results;
+}
+
 int AudioDecoderIlbc::SampleRateHz() const {
   return 8000;
 }
 
 size_t AudioDecoderIlbc::Channels() const {
   return 1;
 }
 
 }  // namespace webrtc