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}
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 a26b2e2..969ff40 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 +291,84 @@
}
}
+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
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index d53551f..5e06d48 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -156,8 +156,8 @@
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.
diff --git a/webrtc/modules/audio_coding/neteq/sync_buffer.h b/webrtc/modules/audio_coding/neteq/sync_buffer.h
index c3bb4a9..5eae4bf 100644
--- a/webrtc/modules/audio_coding/neteq/sync_buffer.h
+++ b/webrtc/modules/audio_coding/neteq/sync_buffer.h
@@ -67,8 +67,7 @@
// 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_|.