Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/split_by_samples.cc

Index: webrtc/modules/audio_coding/codecs/split_by_samples.cc
diff --git a/webrtc/modules/audio_coding/codecs/split_by_samples.cc b/webrtc/modules/audio_coding/codecs/split_by_samples.cc
new file mode 100644
index 0000000000000000000000000000000000000000..6c4f6796f62486dd695ad94240065ac9f3d9f2ba
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/split_by_samples.cc
@@ -0,0 +1,55 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/split_by_samples.h"
+
+namespace webrtc {
+
+namespace internal {
+
+std::vector<AudioDecoder::PacketSplit> SplitBySamples(
+    rtc::ArrayView<const uint8_t> payload,
+    size_t bytes_per_ms,
+    uint32_t timestamps_per_ms) {
+  RTC_DCHECK(payload.data());
+  std::vector<AudioDecoder::PacketSplit> splits;
+  size_t split_size_bytes = payload.size();
+
+  // Find a "chunk size" >= 20 ms and < 40 ms.
+  size_t min_chunk_size = bytes_per_ms * 20;

[kwiberg-webrtc, 2016/09/12 02:11:01]

const

[ossu, 2016/09/12 11:26:37]

Acknowledged.

+  // Reduce the split size by half as long as |split_size_bytes| is at least
+  // twice the minimum chunk size (so that the resulting size is at least as
+  // large as the minimum chunk size).
+  while (split_size_bytes >= 2 * min_chunk_size) {
+    split_size_bytes >>= 1;
+  }
+  uint32_t timestamps_per_chunk = static_cast<uint32_t>(

[kwiberg-webrtc, 2016/09/12 02:11:01]

const

[ossu, 2016/09/12 11:26:37]

Acknowledged.

+      split_size_bytes * timestamps_per_ms / bytes_per_ms);
+  AudioDecoder::PacketSplit split = {0, split_size_bytes, 0};
+  size_t len = payload.size();
+  for (len = payload.size();

[kwiberg-webrtc, 2016/09/12 02:11:01]

You initialize twice.

[ossu, 2016/09/12 11:26:37]

Would you look at that!

+       len >= (2 * split_size_bytes);
+       len -= split_size_bytes) {
+    splits.push_back(split);
+    split.byte_offset += split_size_bytes;
+    split.timestamp_offset += timestamps_per_chunk;
+  }
+
+  if (len > 0) {
+    split.num_bytes = len;
+    splits.push_back(split);
+  }
+
+  return splits;
+}
+
+}  // namespace internal
+
+}  // namespace webrtc