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"
 
 namespace webrtc {
 
 AudioDecoderIlbc::AudioDecoderIlbc() {
   WebRtcIlbcfix_DecoderCreate(&dec_state_);
   WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 
 AudioDecoderIlbc::~AudioDecoderIlbc() {
@@ skipping 18 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_);
 }
 
+AudioDecoder::PacketSplits AudioDecoderIlbc::SplitPacket(
+    const uint8_t* payload,
+    size_t payload_len) const {
+  PacketSplits splits;
+  size_t bytes_per_frame;
+  int timestamps_per_frame;
+  if (payload_len >= 950) {

[kwiberg-webrtc, 2016/08/26 12:39:25]

It's probably best to name the constants:

  constexpr int bytes_in_20ms_frame = 38;
  constexpr int bytes_in_30ms_frame = 50;
  constexpr int frame_bytes_lcm = 950;  // LCM(38, 50) = 950.
  if (payload_len >= frame_bytes_lcm) {

Hmm. Discarding everything that's longer than the lcm is excessive. It should be
enough to discard only those packets that are actually multiples of both 38 and
50.

+    LOG(LS_WARNING) << "AudioDecoderIlbc::SplitPacket: Payload too large";
+    return splits;
+  }
+  if (payload_len % 38 == 0) {
+    // 20 ms frames.
+    bytes_per_frame = 38;
+    timestamps_per_frame = 160;
+  } else if (payload_len % 50 == 0) {
+    // 30 ms frames.
+    bytes_per_frame = 50;
+    timestamps_per_frame = 240;
+  } else {
+    LOG(LS_WARNING) << "AudioDecoderIlbc::SplitPacket: Invalid payload";
+    return splits;
+  }
+
+  if (payload_len == bytes_per_frame) {
+    // Special case. Do not split the payload.
+    return {PacketSplit{0, payload_len, 0}};
+  }

[kwiberg-webrtc, 2016/08/26 12:39:25]

Is this special case really needed? It looks like the loop would handle it just
fine (and if it doesn't, wouldn't it be ending every sequence with a zero-size
PacketSplit?).

[ossu, 2016/08/26 13:05:30]

You're right. This shouldn't be necessary. I'll remove it.

+
+  PacketSplit split = {0, bytes_per_frame, 0};
+  for (size_t len = payload_len; len > 0; len -= bytes_per_frame) {
+    RTC_DCHECK(len >= bytes_per_frame);
+    splits.push_back(split);
+    split.byte_offset += bytes_per_frame;
+    split.timestamp_offset += timestamps_per_frame;
+  }
+
+  return splits;
+}
+
 int AudioDecoderIlbc::SampleRateHz() const {
   return 8000;
 }
 
 size_t AudioDecoderIlbc::Channels() const {
   return 1;
 }
 
 }  // namespace webrtc