From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Sun, 7 Apr 2024 19:32:43 +0200
Subject: Adding upstream version 1:115.7.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 .../libwebrtc/audio/channel_send_unittest.cc       | 113 +++++++++++++++++++++
 1 file changed, 113 insertions(+)
 create mode 100644 third_party/libwebrtc/audio/channel_send_unittest.cc

(limited to 'third_party/libwebrtc/audio/channel_send_unittest.cc')

diff --git a/third_party/libwebrtc/audio/channel_send_unittest.cc b/third_party/libwebrtc/audio/channel_send_unittest.cc
new file mode 100644
index 0000000000..50d8368d4a
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_send_unittest.cc
@@ -0,0 +1,113 @@
+/*
+ *  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/rtc_event_log/rtc_event_log.h"
+#include "api/scoped_refptr.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "call/rtp_transport_controller_send.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 {
+
+constexpr int kRtcpIntervalMs = 1000;
+constexpr int kSsrc = 333;
+constexpr int kPayloadType = 1;
+
+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;
+}
+
+std::unique_ptr<AudioFrame> CreateAudioFrame() {
+  auto frame = std::make_unique<AudioFrame>();
+  frame->samples_per_channel_ = 480;
+  frame->sample_rate_hz_ = 48000;
+  frame->num_channels_ = 1;
+  return frame;
+}
+
+class ChannelSendTest : public ::testing::Test {
+ protected:
+  ChannelSendTest()
+      : time_controller_(Timestamp::Seconds(1)),
+        transport_controller_(
+            time_controller_.GetClock(),
+            RtpTransportConfig{
+                .bitrate_config = GetBitrateConfig(),
+                .event_log = &event_log_,
+                .task_queue_factory = time_controller_.GetTaskQueueFactory(),
+                .trials = &field_trials_,
+            }) {
+    transport_controller_.EnsureStarted();
+  }
+
+  std::unique_ptr<ChannelSendInterface> CreateChannelSend() {
+    return voe::CreateChannelSend(
+        time_controller_.GetClock(), time_controller_.GetTaskQueueFactory(),
+        &transport_, nullptr, &event_log_, nullptr, crypto_options_, false,
+        kRtcpIntervalMs, kSsrc, nullptr, nullptr, field_trials_);
+  }
+
+  GlobalSimulatedTimeController time_controller_;
+  webrtc::test::ScopedKeyValueConfig field_trials_;
+  RtcEventLogNull event_log_;
+  MockTransport transport_;
+  RtpTransportControllerSend transport_controller_;
+  CryptoOptions crypto_options_;
+};
+
+TEST_F(ChannelSendTest, StopSendShouldResetEncoder) {
+  std::unique_ptr<ChannelSendInterface> channel = CreateChannelSend();
+  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory =
+      CreateBuiltinAudioEncoderFactory();
+  std::unique_ptr<AudioEncoder> encoder = encoder_factory->MakeAudioEncoder(
+      kPayloadType, SdpAudioFormat("opus", 48000, 2), {});
+  channel->SetEncoder(kPayloadType, std::move(encoder));
+  channel->RegisterSenderCongestionControlObjects(&transport_controller_,
+                                                  nullptr);
+  channel->StartSend();
+
+  // Insert two frames which should trigger a new packet.
+  EXPECT_CALL(transport_, SendRtp).Times(1);
+  channel->ProcessAndEncodeAudio(CreateAudioFrame());
+  time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
+  channel->ProcessAndEncodeAudio(CreateAudioFrame());
+  time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
+
+  EXPECT_CALL(transport_, SendRtp).Times(0);
+  channel->ProcessAndEncodeAudio(CreateAudioFrame());
+  time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
+  // 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.
+  channel->ProcessAndEncodeAudio(CreateAudioFrame());
+  time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
+}
+
+}  // namespace
+}  // namespace voe
+}  // namespace webrtc
-- 
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