Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
 
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
 using webrtc::NetEq;
 using webrtc::test::AudioLoop;
 using webrtc::test::RtpGenerator;
 using webrtc::WebRtcRTPHeader;
 
 namespace webrtc {
@@ skipping 69 matching lines @@
                                                   &rtp_header);
       input_samples = audio_loop.GetNextBlock();
       if (input_samples.empty())
         return -1;
       payload_len = WebRtcPcm16b_Encode(input_samples.data(),
                                         input_samples.size(), input_payload);
       assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
     }
 
     // Get output audio, but don't do anything with it.
-    static const int kMaxChannels = 1;
-    static const size_t kMaxSamplesPerMs = 48000 / 1000;
-    static const int kOutputBlockSizeMs = 10;
-    static const size_t kOutDataLen =
-        kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
-    int16_t out_data[kOutDataLen];
-    size_t num_channels;
-    size_t samples_per_channel;
-    int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
-                                &num_channels, NULL);
+    AudioFrame out_frame;
+    int error = neteq->GetAudio(&out_frame, NULL);
     if (error != NetEq::kOK)
       return -1;
 
-    assert(samples_per_channel == static_cast<size_t>(kSampRateHz * 10 / 1000));
+    assert(out_frame.samples_per_channel_ ==
+           static_cast<size_t>(kSampRateHz * 10 / 1000));
 
+    static const int kOutputBlockSizeMs = 10;
     time_now_ms += kOutputBlockSizeMs;
     if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
       // Apply negative drift second half of simulation.
       rtp_gen.set_drift_factor(-drift_factor);
       drift_flipped = true;
     }
   }
   int64_t end_time_ms = clock->TimeInMilliseconds();
   delete neteq;
   return end_time_ms - start_time_ms;
 }
 
 }  // namespace test
 }  // namespace webrtc