Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc

 /*
  *  Copyright (c) 2014 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 756 matching lines @@
     return test_complete_->Wait(10 * 60 * 1000);  // 10 minutes' timeout.
   }
 
   static bool CbReceiveThread(void* context) {
     return reinterpret_cast<AcmReRegisterIsacMtTestOldApi*>(context)
         ->CbReceiveImpl();
   }
 
   bool CbReceiveImpl() {
     SleepMs(1);
-    const size_t max_encoded_bytes = isac_encoder_->MaxEncodedBytes();
-    std::unique_ptr<uint8_t[]> encoded(new uint8_t[max_encoded_bytes]);
+    rtc::Buffer encoded;
     AudioEncoder::EncodedInfo info;
     {
       rtc::CritScope lock(&crit_sect_);
       if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
         return true;
       }
       next_insert_packet_time_ms_ += kPacketSizeMs;
       ++receive_packet_count_;
 
       // Encode new frame.
       uint32_t input_timestamp = rtp_header_.header.timestamp;
       while (info.encoded_bytes == 0) {
         info =
             isac_encoder_->Encode(input_timestamp, audio_loop_.GetNextBlock(),
-                                  max_encoded_bytes, encoded.get());
+                                  &encoded);
         input_timestamp += 160;  // 10 ms at 16 kHz.
       }
       EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,
                 input_timestamp);
       EXPECT_EQ(rtp_header_.header.timestamp, info.encoded_timestamp);
       EXPECT_EQ(rtp_header_.header.payloadType, info.payload_type);
     }
     // Now we're not holding the crit sect when calling ACM.
 
     // Insert into ACM.
-    EXPECT_EQ(0, acm_->IncomingPacket(encoded.get(), info.encoded_bytes,
+    EXPECT_EQ(0, acm_->IncomingPacket(encoded.data(), info.encoded_bytes,
                                       rtp_header_));
 
     // Pull audio.
     for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) {
       AudioFrame audio_frame;
       EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */,
                                          &audio_frame));
       fake_clock_->AdvanceTimeMilliseconds(10);
     }
     rtp_utility_->Forward(&rtp_header_);
@@ skipping 812 matching lines @@
 TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
   CodecInst codec_inst;
   codec_inst.channels = 1;
   codec_inst.pacsize = 160;
   codec_inst.pltype = 0;
   AudioEncoderPcmU encoder(codec_inst);
   MockAudioEncoder mock_encoder;
   // Set expectations on the mock encoder and also delegate the calls to the
   // real encoder.
   EXPECT_CALL(mock_encoder, MaxEncodedBytes())
-      .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::MaxEncodedBytes));

[kwiberg-webrtc, 2016/02/25 00:29:04]

Do you still need this one?

[ossu, 2016/02/25 10:39:51]

Hmm... Come to think of it, probably not. I'll try and remove it.

   EXPECT_CALL(mock_encoder, SampleRateHz())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SampleRateHz));
   EXPECT_CALL(mock_encoder, NumChannels())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::NumChannels));
   EXPECT_CALL(mock_encoder, RtpTimestampRateHz())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::RtpTimestampRateHz));
   EXPECT_CALL(mock_encoder, Num10MsFramesInNextPacket())
       .Times(AtLeast(1))
       .WillRepeatedly(
           Invoke(&encoder, &AudioEncoderPcmU::Num10MsFramesInNextPacket));
   EXPECT_CALL(mock_encoder, GetTargetBitrate())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));
-  EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _, _))
+  EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _))
       .Times(AtLeast(1))
-      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::EncodeInternal));
+      .WillRepeatedly(Invoke(&encoder,
+                             static_cast<
+                             AudioEncoder::EncodedInfo(AudioEncoder::*)(
+                                 uint32_t,
+                                 rtc::ArrayView<const int16_t>,
+                                 rtc::Buffer*)>(&AudioEncoderPcmU::Encode)));
   EXPECT_CALL(mock_encoder, SetFec(_))
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SetFec));
   ASSERT_NO_FATAL_FAILURE(
       SetUpTestExternalEncoder(&mock_encoder, codec_inst.pltype));
   Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
       50, test::AcmReceiveTestOldApi::kMonoOutput);
 }
 
 // This test fixture is implemented to run ACM and change the desired output
@@ skipping 109 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc