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..970e32d7d152e6c09e1643f113cf5b8af4ee0c6a 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 +291,55 @@ 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 samples_per_ms = rtc::checked_cast<size_t>(
+      rtc::CheckedDivExact(output_sample_rate_hz, 1000));
+  const int num_10ms_frames = codec.inst.pacsize / (10 * samples_per_ms);

[minyue-webrtc, 2016/04/05 13:00:22]

no CheckedDivExact on this?

[hlundin-webrtc, 2016/04/05 13:24:14]

Done.

+
+  // Expect the first output timestamp to be 30 samples before the first
+  // inserted timestamp (because of NetEq's look-ahead).
+  uint32_t expected_output_ts = last_packet_send_timestamp_ - 30u;

[minyue-webrtc, 2016/04/05 13:00:22]

what would it be if |last_packet_send_timestamp_| is smaller than 30u?

[hlundin-webrtc, 2016/04/05 13:24:14]

In fact it is; last_packet_send_timestamp_ = 0 from the test setup. The
subtraction wraps to the high end of uint32_t as expected.

[minyue-webrtc, 2016/04/05 13:30:04]

Acknowledged.

+
+  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(2u, frame.num_channels_);
+      EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
+      EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_);

[minyue-webrtc, 2016/04/05 13:00:22]

why is VadActive when packet is silent?

[hlundin-webrtc, 2016/04/05 13:24:14]

There is a restriction on the post-decode VAD in NetEq that makes it always say
"active speech" for sample rates > 16000 Hz. Legacy...

[minyue-webrtc, 2016/04/05 13:30:04]

so if codec_id equals to a 16kHz codec, this line will fail?

Would you add a comment? or even drop this line.

+    }
+  }
+}
+
 #if defined(WEBRTC_ANDROID)
 #define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
 #else