Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h

Index: webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
diff --git a/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
new file mode 100644
index 0000000000000000000000000000000000000000..5c080c5ce3434ffe04ac710a471591febd2bbc99
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+
+namespace webrtc {
+
+class LegacyEncodedAudioFrame : public AudioDecoder::EncodedAudioFrame {
+ public:
+  LegacyEncodedAudioFrame(AudioDecoder* decoder,
+                          rtc::Buffer* payload,
+                          bool is_primary_payload);
+  ~LegacyEncodedAudioFrame() override;
+
+  static std::vector<AudioDecoder::ParseResult> SplitBySamples(

[ossu, 2016/09/13 14:28:29]

I've kept this as SplitBySamples for now, but it could have a more descriptive
name, such as: CreateFromSampleBasedPayload or something.

[hlundin-webrtc, 2016/09/15 08:49:14]

I think SplitBySamples is good enough.

[ossu, 2016/09/15 14:41:05]

Acknowledged.

+      AudioDecoder* decoder,
+      rtc::Buffer* payload,
+      uint32_t timestamp,
+      bool is_primary,
+      size_t bytes_per_ms,
+      uint32_t timestamps_per_ms);
+
+  size_t Duration() const override;
+
+  rtc::Optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override;
+
+  // For testing:
+  const rtc::Buffer& payload() const { return payload_; }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+  const bool is_primary_payload_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_