Moved codec-specific audio packet splitting into decoders.

webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc

Index: webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
index dab5805f98afcc41eaa714d49e4cd8baec5ebe59..9eacfc0ff0ac24fa8d221c2759eebda2bcf65c0b 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
@@ -11,7 +11,9 @@
 #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 {
 
@@ -49,6 +51,51 @@ 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::SplitPacket: Payload too large";

[ossu, 2016/09/13 14:25:55]

This comment should say ParsePayload and not SplitPacket.

+    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::SplitPacket: Invalid payload";

[ossu, 2016/09/13 14:25:55]

This comment should say ParsePayload and not SplitPacket.

+    return results;
+  }
+
+  RTC_DCHECK(payload->size() % bytes_per_frame == 0);

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

RTC_DCHECK_EQ

[ossu, 2016/09/15 08:58:11]

I did that initially, but the RTC_DHECK_EQ macro barfed on me, whining about
comparing a size_t (payload->size() % bytes_per_frame) to an int (0). I could
get around it by typing the zero, but it's a bit longer. I'll do that, though,
so the DCHECK message printout is a bit better.

[hlundin-webrtc, 2016/09/15 09:43:38]

I hate it when that happens, but I prefer the nice printouts.

[kwiberg-webrtc, 2016/09/15 13:01:25]

+1. It's annoying, but since a "u" suffix will solve the problem, I've never
tried to fix it for real.

+  if (payload->size() == bytes_per_frame) {
+    std::unique_ptr<EncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(this, payload, is_primary));
+    results.emplace_back(timestamp, is_primary, std::move(frame));
+  } else {
+    for (size_t 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, &new_payload, is_primary));
+      results.emplace_back(timestamp + timestamp_offset, is_primary,
+                           std::move(frame));
+    }
+  }
+
+  return results;
+}
+
 int AudioDecoderIlbc::SampleRateHz() const {
   return 8000;
 }