summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/audio/channel_send_unittest.cc
blob: 77d847951937c0d4aeb71a163c73d3a948c8ffcc (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
/*
 *  Copyright 2023 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "audio/channel_send.h"

#include <utility>

#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/scoped_refptr.h"
#include "api/test/mock_frame_transformer.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "call/rtp_transport_controller_send.h"
#include "rtc_base/gunit.h"
#include "test/gtest.h"
#include "test/mock_transport.h"
#include "test/scoped_key_value_config.h"
#include "test/time_controller/simulated_time_controller.h"

namespace webrtc {
namespace voe {
namespace {

using ::testing::Invoke;
using ::testing::NiceMock;
using ::testing::Return;
using ::testing::SaveArg;

constexpr int kRtcpIntervalMs = 1000;
constexpr int kSsrc = 333;
constexpr int kPayloadType = 1;
constexpr int kSampleRateHz = 48000;
constexpr int kRtpRateHz = 48000;

BitrateConstraints GetBitrateConfig() {
  BitrateConstraints bitrate_config;
  bitrate_config.min_bitrate_bps = 10000;
  bitrate_config.start_bitrate_bps = 100000;
  bitrate_config.max_bitrate_bps = 1000000;
  return bitrate_config;
}

class ChannelSendTest : public ::testing::Test {
 protected:
  ChannelSendTest()
      : time_controller_(Timestamp::Seconds(1)),
        env_(CreateEnvironment(&field_trials_,
                               time_controller_.GetClock(),
                               time_controller_.CreateTaskQueueFactory())),
        transport_controller_(
            RtpTransportConfig{.env = env_,
                               .bitrate_config = GetBitrateConfig()}) {
    channel_ = voe::CreateChannelSend(
        time_controller_.GetClock(), time_controller_.GetTaskQueueFactory(),
        &transport_, nullptr, &env_.event_log(), nullptr, crypto_options_,
        false, kRtcpIntervalMs, kSsrc, nullptr, &transport_controller_,
        env_.field_trials());
    encoder_factory_ = CreateBuiltinAudioEncoderFactory();
    SdpAudioFormat opus = SdpAudioFormat("opus", kRtpRateHz, 2);
    std::unique_ptr<AudioEncoder> encoder =
        encoder_factory_->MakeAudioEncoder(kPayloadType, opus, {});
    channel_->SetEncoder(kPayloadType, opus, std::move(encoder));
    transport_controller_.EnsureStarted();
    channel_->RegisterSenderCongestionControlObjects(&transport_controller_);
    ON_CALL(transport_, SendRtcp).WillByDefault(Return(true));
    ON_CALL(transport_, SendRtp).WillByDefault(Return(true));
  }

  std::unique_ptr<AudioFrame> CreateAudioFrame() {
    auto frame = std::make_unique<AudioFrame>();
    frame->sample_rate_hz_ = kSampleRateHz;
    frame->samples_per_channel_ = kSampleRateHz / 100;
    frame->num_channels_ = 1;
    frame->set_absolute_capture_timestamp_ms(
        time_controller_.GetClock()->TimeInMilliseconds());
    return frame;
  }

  void ProcessNextFrame() {
    channel_->ProcessAndEncodeAudio(CreateAudioFrame());
    // Advance time to process the task queue.
    time_controller_.AdvanceTime(TimeDelta::Millis(10));
  }

  GlobalSimulatedTimeController time_controller_;
  webrtc::test::ScopedKeyValueConfig field_trials_;
  Environment env_;
  NiceMock<MockTransport> transport_;
  CryptoOptions crypto_options_;
  RtpTransportControllerSend transport_controller_;
  std::unique_ptr<ChannelSendInterface> channel_;
  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
};

TEST_F(ChannelSendTest, StopSendShouldResetEncoder) {
  channel_->StartSend();
  // Insert two frames which should trigger a new packet.
  EXPECT_CALL(transport_, SendRtp).Times(1);
  ProcessNextFrame();
  ProcessNextFrame();

  EXPECT_CALL(transport_, SendRtp).Times(0);
  ProcessNextFrame();
  // StopSend should clear the previous audio frame stored in the encoder.
  channel_->StopSend();
  channel_->StartSend();
  // The following frame should not trigger a new packet since the encoder
  // needs 20 ms audio.
  EXPECT_CALL(transport_, SendRtp).Times(0);
  ProcessNextFrame();
}

TEST_F(ChannelSendTest, IncreaseRtpTimestampByPauseDuration) {
  channel_->StartSend();
  uint32_t timestamp;
  int sent_packets = 0;
  auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
                      const PacketOptions& options) {
    ++sent_packets;
    RtpPacketReceived packet;
    packet.Parse(data);
    timestamp = packet.Timestamp();
    return true;
  };
  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(send_rtp));
  ProcessNextFrame();
  ProcessNextFrame();
  EXPECT_EQ(sent_packets, 1);
  uint32_t first_timestamp = timestamp;
  channel_->StopSend();
  time_controller_.AdvanceTime(TimeDelta::Seconds(10));
  channel_->StartSend();

  ProcessNextFrame();
  ProcessNextFrame();
  EXPECT_EQ(sent_packets, 2);
  int64_t timestamp_gap_ms =
      static_cast<int64_t>(timestamp - first_timestamp) * 1000 / kRtpRateHz;
  EXPECT_EQ(timestamp_gap_ms, 10020);
}

TEST_F(ChannelSendTest, FrameTransformerGetsCorrectTimestamp) {
  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
      rtc::make_ref_counted<MockFrameTransformer>();
  channel_->SetEncoderToPacketizerFrameTransformer(mock_frame_transformer);
  rtc::scoped_refptr<TransformedFrameCallback> callback;
  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
      .WillOnce(SaveArg<0>(&callback));
  EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);

  absl::optional<uint32_t> sent_timestamp;
  auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
                      const PacketOptions& options) {
    RtpPacketReceived packet;
    packet.Parse(data);
    if (!sent_timestamp) {
      sent_timestamp = packet.Timestamp();
    }
    return true;
  };
  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(send_rtp));

  channel_->StartSend();
  int64_t transformable_frame_timestamp = -1;
  EXPECT_CALL(*mock_frame_transformer, Transform)
      .WillOnce([&](std::unique_ptr<TransformableFrameInterface> frame) {
        transformable_frame_timestamp = frame->GetTimestamp();
        callback->OnTransformedFrame(std::move(frame));
      });
  // Insert two frames which should trigger a new packet.
  ProcessNextFrame();
  ProcessNextFrame();

  // Ensure the RTP timestamp on the frame passed to the transformer
  // includes the RTP offset and matches the actual RTP timestamp on the sent
  // packet.
  EXPECT_EQ_WAIT(transformable_frame_timestamp,
                 0 + channel_->GetRtpRtcp()->StartTimestamp(), 1000);
  EXPECT_TRUE_WAIT(sent_timestamp, 1000);
  EXPECT_EQ(*sent_timestamp, transformable_frame_timestamp);
}
}  // namespace
}  // namespace voe
}  // namespace webrtc