Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/audio_decoder.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 47 matching lines @@
 }
 
 int AudioDecoder::IncomingPacket(const uint8_t* payload,
                                  size_t payload_len,
                                  uint16_t rtp_sequence_number,
                                  uint32_t rtp_timestamp,
                                  uint32_t arrival_timestamp) {
   return 0;
 }
 
+AudioDecoder::PacketSplits AudioDecoder::SplitPacket(const uint8_t* payload,
+                                                     size_t payload_len) const {
+  return {PacketSplit{0, payload_len, 0}};
+}
+
 int AudioDecoder::ErrorCode() { return 0; }
 
 int AudioDecoder::PacketDuration(const uint8_t* encoded,
                                  size_t encoded_len) const {
   return kNotImplemented;
 }
 
 int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
                                           size_t encoded_len) const {
   return kNotImplemented;
@@ skipping 10 matching lines @@
     case 1:
       return kSpeech;
     case 2:
       return kComfortNoise;
     default:
       assert(false);
       return kSpeech;
   }
 }
 
+AudioDecoder::PacketSplits AudioDecoder::SplitBySamples(
+    const uint8_t* payload,
+    size_t payload_len,
+    size_t bytes_per_ms,
+    uint32_t timestamps_per_ms) {

[kwiberg-webrtc, 2016/08/26 12:39:24]

This is the sort of function that should have a unit test.

[ossu, 2016/08/26 13:05:29]

It does. Several of them even.
Though, granted, it's not tested by calling SplitBySamples directly, but the
same way it was tested before: by setting up codecs with known splits and
testing the splitting on them.

[kwiberg-webrtc, 2016/08/26 22:05:13]

It's arguably not a unit test of this function in that case. :-) But that's
certainly much better than nothing.

+  PacketSplits splits;
+  RTC_DCHECK(payload);
+
+  size_t split_size_bytes = payload_len;
+
+  // Find a "chunk size" >= 20 ms and < 40 ms.
+  size_t min_chunk_size = bytes_per_ms * 20;

[kwiberg-webrtc, 2016/08/26 12:39:24]

const

[ossu, 2016/08/26 13:05:29]

Hey, don't blame me, I just clean up around here! :)

[kwiberg-webrtc, 2016/08/26 22:05:13]

Yeah, I think I wrote this comment before I realized that you'd just moved this
code from somewhere else.

+  // 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;

[kwiberg-webrtc, 2016/08/26 12:39:24]

Less cleverly written as split_size_bytes /= 2;

I'm sure there's a way to do this without looping, but we can leave that
optimization for another day...

[ossu, 2016/08/26 13:05:29]

Yeah, I mean the code can be improved but I've aimed to change as little as
possible during the restructuring. I could simplify it in a separate CL and if
something breaks, we know it's not due to moving stuff around, but due to the
cleanup.

[kwiberg-webrtc, 2016/08/26 22:05:13]

sgtm.

+  }
+  uint32_t timestamps_per_chunk = static_cast<uint32_t>(

[kwiberg-webrtc, 2016/08/26 12:39:24]

const

+      split_size_bytes * timestamps_per_ms / bytes_per_ms);
+  PacketSplit split = {0, split_size_bytes, 0};
+  size_t len = payload_len;
+  for (len = payload_len;
+       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 webrtc