1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include <memory>
12
13 #include "absl/flags/flag.h"
14 #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
15 #include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
16 #include "rtc_base/checks.h"
17 #include "rtc_base/numerics/safe_conversions.h"
18 #include "test/testsupport/file_utils.h"
19
20 ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
21
22 using ::testing::InitGoogleTest;
23
24 namespace webrtc {
25 namespace test {
26 namespace {
27 static const int kInputSampleRateKhz = 8;
28 static const int kOutputSampleRateKhz = 8;
29 } // namespace
30
31 class NetEqPcmuQualityTest : public NetEqQualityTest {
32 protected:
NetEqPcmuQualityTest()33 NetEqPcmuQualityTest()
34 : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
35 kInputSampleRateKhz,
36 kOutputSampleRateKhz,
37 SdpAudioFormat("pcmu", 8000, 1)) {
38 // Flag validation
39 RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
40 absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
41 (absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
42 << "Invalid frame size, should be 10, 20, ..., 60 ms.";
43 }
44
SetUp()45 void SetUp() override {
46 ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
47 AudioEncoderPcmU::Config config;
48 config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
49 encoder_.reset(new AudioEncoderPcmU(config));
50 NetEqQualityTest::SetUp();
51 }
52
EncodeBlock(int16_t * in_data,size_t block_size_samples,rtc::Buffer * payload,size_t max_bytes)53 int EncodeBlock(int16_t* in_data,
54 size_t block_size_samples,
55 rtc::Buffer* payload,
56 size_t max_bytes) override {
57 const size_t kFrameSizeSamples = 80; // Samples per 10 ms.
58 size_t encoded_samples = 0;
59 uint32_t dummy_timestamp = 0;
60 AudioEncoder::EncodedInfo info;
61 do {
62 info = encoder_->Encode(dummy_timestamp,
63 rtc::ArrayView<const int16_t>(
64 in_data + encoded_samples, kFrameSizeSamples),
65 payload);
66 encoded_samples += kFrameSizeSamples;
67 } while (info.encoded_bytes == 0);
68 return rtc::checked_cast<int>(info.encoded_bytes);
69 }
70
71 private:
72 std::unique_ptr<AudioEncoderPcmU> encoder_;
73 };
74
TEST_F(NetEqPcmuQualityTest,Test)75 TEST_F(NetEqPcmuQualityTest, Test) {
76 Simulate();
77 }
78
79 } // namespace test
80 } // namespace webrtc
81