Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 161 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receive_test.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receive_test.cc
new file mode 100644
index 0000000000..66f6255b01
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receive_test.cc
@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2014 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 "modules/audio_coding/acm2/acm_receive_test.h"
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+acm2::AcmReceiver::Config MakeAcmConfig(
+    Clock& clock,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
+  acm2::AcmReceiver::Config config;
+  config.clock = clock;
+  config.decoder_factory = std::move(decoder_factory);
+  return config;
+}
+}  // namespace
+
+AcmReceiveTestOldApi::AcmReceiveTestOldApi(
+    PacketSource* packet_source,
+    AudioSink* audio_sink,
+    int output_freq_hz,
+    NumOutputChannels exptected_output_channels,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
+    : clock_(0),
+      acm_receiver_(std::make_unique<acm2::AcmReceiver>(
+          MakeAcmConfig(clock_, std::move(decoder_factory)))),
+      packet_source_(packet_source),
+      audio_sink_(audio_sink),
+      output_freq_hz_(output_freq_hz),
+      exptected_output_channels_(exptected_output_channels) {}
+
+AcmReceiveTestOldApi::~AcmReceiveTestOldApi() = default;
+
+void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
+  acm_receiver_->SetCodecs({{103, {"ISAC", 16000, 1}},
+                            {104, {"ISAC", 32000, 1}},
+                            {107, {"L16", 8000, 1}},
+                            {108, {"L16", 16000, 1}},
+                            {109, {"L16", 32000, 1}},
+                            {111, {"L16", 8000, 2}},
+                            {112, {"L16", 16000, 2}},
+                            {113, {"L16", 32000, 2}},
+                            {0, {"PCMU", 8000, 1}},
+                            {110, {"PCMU", 8000, 2}},
+                            {8, {"PCMA", 8000, 1}},
+                            {118, {"PCMA", 8000, 2}},
+                            {102, {"ILBC", 8000, 1}},
+                            {9, {"G722", 8000, 1}},
+                            {119, {"G722", 8000, 2}},
+                            {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+                            {13, {"CN", 8000, 1}},
+                            {98, {"CN", 16000, 1}},
+                            {99, {"CN", 32000, 1}}});
+}
+
+// Remaps payload types from ACM's default to those used in the resource file
+// neteq_universal_new.rtp.
+void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
+  acm_receiver_->SetCodecs({{103, {"ISAC", 16000, 1}},
+                            {104, {"ISAC", 32000, 1}},
+                            {93, {"L16", 8000, 1}},
+                            {94, {"L16", 16000, 1}},
+                            {95, {"L16", 32000, 1}},
+                            {0, {"PCMU", 8000, 1}},
+                            {8, {"PCMA", 8000, 1}},
+                            {102, {"ILBC", 8000, 1}},
+                            {9, {"G722", 8000, 1}},
+                            {120, {"OPUS", 48000, 2}},
+                            {13, {"CN", 8000, 1}},
+                            {98, {"CN", 16000, 1}},
+                            {99, {"CN", 32000, 1}}});
+}
+
+void AcmReceiveTestOldApi::Run() {
+  for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
+       packet = packet_source_->NextPacket()) {
+    // Pull audio until time to insert packet.
+    while (clock_.TimeInMilliseconds() < packet->time_ms()) {
+      AudioFrame output_frame;
+      bool muted;
+      EXPECT_EQ(
+          0, acm_receiver_->GetAudio(output_freq_hz_, &output_frame, &muted));
+      ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
+      ASSERT_FALSE(muted);
+      const size_t samples_per_block =
+          static_cast<size_t>(output_freq_hz_ * 10 / 1000);
+      EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
+      if (exptected_output_channels_ != kArbitraryChannels) {
+        if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
+          // Don't check number of channels for PLC output, since each test run
+          // usually starts with a short period of mono PLC before decoding the
+          // first packet.
+        } else {
+          EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
+        }
+      }
+      ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
+      clock_.AdvanceTimeMilliseconds(10);
+      AfterGetAudio();
+    }
+
+    EXPECT_EQ(0, acm_receiver_->InsertPacket(
+                     packet->header(),
+                     rtc::ArrayView<const uint8_t>(
+                         packet->payload(), packet->payload_length_bytes())))
+        << "Failure when inserting packet:" << std::endl
+        << "  PT = " << static_cast<int>(packet->header().payloadType)
+        << std::endl
+        << "  TS = " << packet->header().timestamp << std::endl
+        << "  SN = " << packet->header().sequenceNumber;
+  }
+}
+
+AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
+    PacketSource* packet_source,
+    AudioSink* audio_sink,
+    int output_freq_hz_1,
+    int output_freq_hz_2,
+    int toggle_period_ms,
+    NumOutputChannels exptected_output_channels)
+    : AcmReceiveTestOldApi(packet_source,
+                           audio_sink,
+                           output_freq_hz_1,
+                           exptected_output_channels,
+                           CreateBuiltinAudioDecoderFactory()),
+      output_freq_hz_1_(output_freq_hz_1),
+      output_freq_hz_2_(output_freq_hz_2),
+      toggle_period_ms_(toggle_period_ms),
+      last_toggle_time_ms_(clock_.TimeInMilliseconds()) {}
+
+void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
+  if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
+    output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
+                          ? output_freq_hz_2_
+                          : output_freq_hz_1_;
+    last_toggle_time_ms_ = clock_.TimeInMilliseconds();
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc