Unit test for AudioFrame output from AcmReceiver::GetAudio

webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc

Index: webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
index a26b2e217fe2a90a13c4e0389f2dc6807ecbb798..bc95edaa5ca159c0a428353e8c5d7485fb3312b0 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
@@ -14,6 +14,8 @@
 #include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
@@ -289,6 +291,63 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
   }
 }
 
+class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
+ protected:
+  AcmReceiverTestFaxModeOldApi() {
+    config_.neteq_config.playout_mode = kPlayoutFax;
+  }
+};
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyAudioFrame DISABLED_VerifyAudioFrame
+#else
+#define MAYBE_VerifyAudioFrame VerifyAudioFrame
+#endif
+TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrame) {
+  // Make sure "fax mode" is enabled. This will avoid delay changes unless the
+  // packet-loss concealment is made. We do this in order to make the timestamp
+  // increments predictable; in normal mode, NetEq may decide to do accelerate
+  // or pre-emptive expand operations after some time, offsetting the timestamp.
+  EXPECT_EQ(kPlayoutFax, config_.neteq_config.playout_mode);
+
+  const RentACodec::CodecId codec_id = RentACodec::CodecId::kOpus;
+  const RentACodec::CodecId kCodecId[] = {codec_id};
+  AddSetOfCodecs(kCodecId);
+
+  const CodecIdInst codec(codec_id);
+  const int output_sample_rate_hz = codec.inst.plfreq;
+  const size_t output_channels = codec.inst.channels;
+  const size_t samples_per_ms = rtc::checked_cast<size_t>(
+      rtc::CheckedDivExact(output_sample_rate_hz, 1000));
+  const int num_10ms_frames = rtc::CheckedDivExact(
+      codec.inst.pacsize, rtc::checked_cast<int>(10 * samples_per_ms));
+  const AudioFrame::VADActivity expected_vad_activity =
+      output_sample_rate_hz > 16000 ? AudioFrame::kVadActive
+                                    : AudioFrame::kVadPassive;
+
+  // Expect the first output timestamp to be 5*fs/8000 samples before the first
+  // inserted timestamp (because of NetEq's look-ahead). (This value is defined
+  // in Expand::overlap_length_.)
+  uint32_t expected_output_ts =
+      last_packet_send_timestamp_ -

[minyue-webrtc, 2016/04/05 14:08:59]

better moving this line up

[hlundin-webrtc, 2016/04/05 14:24:15]

Done.

+      rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);
+
+  AudioFrame frame;
+  for (int i = 0; i < 5; ++i) {
+    InsertOnePacketOfSilence(codec.id);
+    for (int k = 0; k < num_10ms_frames; ++k) {
+      EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame));
+      EXPECT_EQ(expected_output_ts, frame.timestamp_);
+      expected_output_ts += 10 * samples_per_ms;
+      EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
+      EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
+      EXPECT_EQ(output_channels, frame.num_channels_);
+      EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
+      EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
+    }
+  }
+}
+
 #if defined(WEBRTC_ANDROID)
 #define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
 #else