Unit test for AudioFrame output from AcmReceiver::GetAudio

This new unit test verifies the parameter fields (not the audio data
itself) written to the AudioFrame output by AcmReceiver::GetAudio.

Also corrected a few comments reflecting recent changes in the code.

BUG=webrtc:5669

Review URL: https://codereview.webrtc.org/1859953002

Cr-Commit-Position: refs/heads/master@{#12253}

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 +291,84 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
   }
 }
 
+class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
+ protected:
+  AcmReceiverTestFaxModeOldApi() {
+    config_.neteq_config.playout_mode = kPlayoutFax;
+  }
+
+  void RunVerifyAudioFrame(RentACodec::CodecId codec_id) {
+    // 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 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_ -
+        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_VerifyAudioFramePCMU DISABLED_VerifyAudioFramePCMU
+#else
+#define MAYBE_VerifyAudioFramePCMU VerifyAudioFramePCMU
+#endif
+TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFramePCMU) {
+  RunVerifyAudioFrame(RentACodec::CodecId::kPCMU);
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyAudioFrameISAC DISABLED_VerifyAudioFrameISAC
+#else
+#define MAYBE_VerifyAudioFrameISAC VerifyAudioFrameISAC
+#endif
+TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameISAC) {
+  RunVerifyAudioFrame(RentACodec::CodecId::kISAC);
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyAudioFrameOpus DISABLED_VerifyAudioFrameOpus
+#else
+#define MAYBE_VerifyAudioFrameOpus VerifyAudioFrameOpus
+#endif
+TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameOpus) {
+  RunVerifyAudioFrame(RentACodec::CodecId::kOpus);
+}
+
 #if defined(WEBRTC_ANDROID)
 #define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
 #else

webrtc/modules/audio_coding/neteq/include/neteq.h

@@ -156,8 +156,8 @@ class NetEq {
                                uint32_t receive_timestamp) = 0;
 
   // Instructs NetEq to deliver 10 ms of audio data. The data is written to
-  // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |interleaved_|,
-  // |num_channels_|, |samples_per_channel_|, |speech_type_|, and
+  // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |speech_type_|,
+  // |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and
   // |vad_activity_| are updated upon success. If an error is returned, some
   // fields may not have been updated.
   // Returns kOK on success, or kFail in case of an error.

webrtc/modules/audio_coding/neteq/sync_buffer.h

@@ -67,8 +67,7 @@ class SyncBuffer : public AudioMultiVector {
   // Reads |requested_len| samples from each channel and writes them interleaved
   // into |output|. The |next_index_| is updated to point to the sample to read
   // next time. The AudioFrame |output| is first reset, and the |data_|,
-  // |interleaved_|, |num_channels_|, and |samples_per_channel_| fields are
-  // updated.
+  // |num_channels_|, and |samples_per_channel_| fields are updated.
   void GetNextAudioInterleaved(size_t requested_len, AudioFrame* output);
 
   // Adds |increment| to |end_timestamp_|.