Make FakeDecodeFromFile handle codec-internal CNG

webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc

 /*
  *  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/neteq/tools/fake_decode_from_file.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 
 namespace webrtc {
 namespace test {
 
 int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
                                        size_t encoded_len,
-                                       int /*sample_rate_hz*/,
+                                       int sample_rate_hz,
                                        int16_t* decoded,
                                        SpeechType* speech_type) {
-  RTC_CHECK_GE(encoded_len, 8u);
+  if (encoded_len == 0) {
+    // Decoder is asked to produce codec-internal comfort noise.
+    RTC_DCHECK(!encoded);  // NetEq always sends nullptr in this case.
+    RTC_DCHECK(cng_mode_);
+    RTC_DCHECK_GT(last_decoded_length_, 0u);
+    std::fill_n(decoded, last_decoded_length_, 0);
+    *speech_type = kComfortNoise;
+    return last_decoded_length_;
+  }
+
+  RTC_CHECK_GE(encoded_len, 12u);
   uint32_t timestamp_to_decode =
       ByteReader<uint32_t>::ReadLittleEndian(encoded);
   uint32_t samples_to_decode =
       ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
+  if (samples_to_decode == 0) {
+    // Number of samples in packet is unknown.
+    if (last_decoded_length_ > 0) {
+      // Use length of last decoded packet, but since this is the total for all
+      // channels, we have to divide by 2 in the stereo case.
+      samples_to_decode =
+          rtc::CheckedDivExact(last_decoded_length_, (stereo_ ? 2uL : 1uL));
+    } else {
+      // This is the first packet to decode, and we do not know the length of
+      // it. Set it to 10 ms.
+      samples_to_decode = rtc::CheckedDivExact(sample_rate_hz, 100);
+    }
+  }
+  uint32_t original_payload_size_bytes =
+      ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);

[ivoc, 2016/08/24 14:55:28]

I think it makes sense to move this below to just before it is used in the if
statement.

[hlundin-webrtc, 2016/08/24 15:27:36]

Done.

 
   if (next_timestamp_from_input_ &&
       timestamp_to_decode != *next_timestamp_from_input_) {
     // A gap in the timestamp sequence is detected. Skip the same number of
     // samples from the file.
     uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_;
     RTC_CHECK(input_->Seek(jump));
   }
 
-  RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded));
   next_timestamp_from_input_ =
       rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode);
 
+  if (original_payload_size_bytes == 1) {
+    // This is a comfort noise payload.
+    RTC_DCHECK_GT(last_decoded_length_, 0u);
+    std::fill_n(decoded, last_decoded_length_, 0);
+    *speech_type = kComfortNoise;
+    cng_mode_ = true;
+    return last_decoded_length_;
+  }
+
+  cng_mode_ = false;
+  RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded));
+
   if (stereo_) {
     InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2,
                                          decoded);
     samples_to_decode *= 2;
   }
 
   *speech_type = kSpeech;
-  return samples_to_decode;
+  return last_decoded_length_ = samples_to_decode;
 }
 
 void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
                                         size_t samples,
+                                        size_t original_payload_size_bytes,
                                         rtc::ArrayView<uint8_t> encoded) {
-  RTC_CHECK_GE(encoded.size(), 8u);
+  RTC_CHECK_GE(encoded.size(), 12u);
   ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
   ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
                                           rtc::checked_cast<uint32_t>(samples));
+  ByteWriter<uint32_t>::WriteLittleEndian(
+      &encoded[8], rtc::checked_cast<uint32_t>(original_payload_size_bytes));
 }
 
 }  // namespace test
 }  // namespace webrtc