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47ed99872d
This allows callers to use timestamps generated from their own clocks without worrying about converting to webrtc time. No-Try because of lack of infra lack of capacity on macs. No-Try: True Bug: webrtc:11327 Change-Id: I7b1935654a2b23cf844c7b3622ed68763ced9da5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219785 Commit-Queue: Paul Hallak <phallak@google.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34076}
262 lines
11 KiB
C++
262 lines
11 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
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#include <memory>
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#include <vector>
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#include "api/transport/field_trial_based_config.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "test/field_trial.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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enum : int { // The first valid value is 1.
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kAudioLevelExtensionId = 1,
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kAbsoluteCaptureTimeExtensionId = 2,
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};
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const uint16_t kSeqNum = 33;
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const uint32_t kSsrc = 725242;
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const uint8_t kAudioLevel = 0x5a;
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const uint64_t kStartTime = 123456789;
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using ::testing::ElementsAreArray;
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class LoopbackTransportTest : public webrtc::Transport {
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public:
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LoopbackTransportTest() {
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receivers_extensions_.Register<AudioLevel>(kAudioLevelExtensionId);
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receivers_extensions_.Register<AbsoluteCaptureTimeExtension>(
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kAbsoluteCaptureTimeExtensionId);
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}
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bool SendRtp(const uint8_t* data,
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size_t len,
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const PacketOptions& /*options*/) override {
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sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
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EXPECT_TRUE(sent_packets_.back().Parse(data, len));
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return true;
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}
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bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
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const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); }
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int packets_sent() { return sent_packets_.size(); }
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private:
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RtpHeaderExtensionMap receivers_extensions_;
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std::vector<RtpPacketReceived> sent_packets_;
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};
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} // namespace
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class RtpSenderAudioTest : public ::testing::Test {
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public:
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RtpSenderAudioTest()
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: fake_clock_(kStartTime),
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rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
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RtpRtcpInterface::Configuration config;
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config.audio = true;
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config.clock = &fake_clock_;
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config.outgoing_transport = &transport_;
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config.local_media_ssrc = kSsrc;
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return config;
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}())),
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rtp_sender_audio_(
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std::make_unique<RTPSenderAudio>(&fake_clock_,
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rtp_module_->RtpSender())) {
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rtp_module_->SetSequenceNumber(kSeqNum);
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}
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SimulatedClock fake_clock_;
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LoopbackTransportTest transport_;
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std::unique_ptr<ModuleRtpRtcpImpl2> rtp_module_;
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std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
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};
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TEST_F(RtpSenderAudioTest, SendAudio) {
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const char payload_name[] = "PAYLOAD_NAME";
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const uint8_t payload_type = 127;
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ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
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payload_name, payload_type, 48000, 0, 1500));
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uint8_t payload[] = {47, 11, 32, 93, 89};
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ASSERT_TRUE(
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rtp_sender_audio_->SendAudio(AudioFrameType::kAudioFrameCN, payload_type,
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4321, payload, sizeof(payload),
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/*absolute_capture_timestamp_ms=*/0));
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auto sent_payload = transport_.last_sent_packet().payload();
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EXPECT_THAT(sent_payload, ElementsAreArray(payload));
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}
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TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
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EXPECT_EQ(0, rtp_sender_audio_->SetAudioLevel(kAudioLevel));
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rtp_module_->RegisterRtpHeaderExtension(AudioLevel::kUri,
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kAudioLevelExtensionId);
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const char payload_name[] = "PAYLOAD_NAME";
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const uint8_t payload_type = 127;
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ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
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payload_name, payload_type, 48000, 0, 1500));
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uint8_t payload[] = {47, 11, 32, 93, 89};
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ASSERT_TRUE(
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rtp_sender_audio_->SendAudio(AudioFrameType::kAudioFrameCN, payload_type,
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4321, payload, sizeof(payload),
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/*absolute_capture_timestamp_ms=*/0));
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auto sent_payload = transport_.last_sent_packet().payload();
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EXPECT_THAT(sent_payload, ElementsAreArray(payload));
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// Verify AudioLevel extension.
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bool voice_activity;
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uint8_t audio_level;
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EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>(
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&voice_activity, &audio_level));
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EXPECT_EQ(kAudioLevel, audio_level);
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EXPECT_FALSE(voice_activity);
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}
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TEST_F(RtpSenderAudioTest, SendAudioWithoutAbsoluteCaptureTime) {
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constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
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const char payload_name[] = "audio";
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const uint8_t payload_type = 127;
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ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
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payload_name, payload_type, 48000, 0, 1500));
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uint8_t payload[] = {47, 11, 32, 93, 89};
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ASSERT_TRUE(rtp_sender_audio_->SendAudio(
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AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
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sizeof(payload), kAbsoluteCaptureTimestampMs));
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EXPECT_FALSE(transport_.last_sent_packet()
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.HasExtension<AbsoluteCaptureTimeExtension>());
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}
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TEST_F(RtpSenderAudioTest, SendAudioWithAbsoluteCaptureTime) {
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rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::kUri,
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kAbsoluteCaptureTimeExtensionId);
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constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
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const char payload_name[] = "audio";
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const uint8_t payload_type = 127;
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ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
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payload_name, payload_type, 48000, 0, 1500));
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uint8_t payload[] = {47, 11, 32, 93, 89};
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ASSERT_TRUE(rtp_sender_audio_->SendAudio(
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AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
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sizeof(payload), kAbsoluteCaptureTimestampMs));
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auto absolute_capture_time =
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transport_.last_sent_packet()
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.GetExtension<AbsoluteCaptureTimeExtension>();
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EXPECT_TRUE(absolute_capture_time);
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EXPECT_EQ(
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absolute_capture_time->absolute_capture_timestamp,
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Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
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kAbsoluteCaptureTimestampMs)));
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EXPECT_FALSE(
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absolute_capture_time->estimated_capture_clock_offset.has_value());
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}
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// Essentially the same test as SendAudioWithAbsoluteCaptureTime but with a
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// field trial. After the field trial is experimented, we will remove
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// SendAudioWithAbsoluteCaptureTime.
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TEST_F(RtpSenderAudioTest,
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SendAudioWithAbsoluteCaptureTimeWithCaptureClockOffset) {
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// Recreate rtp_sender_audio_ wieh new field trial.
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test::ScopedFieldTrials field_trial(
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"WebRTC-IncludeCaptureClockOffset/Enabled/");
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rtp_sender_audio_ =
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std::make_unique<RTPSenderAudio>(&fake_clock_, rtp_module_->RtpSender());
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rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::kUri,
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kAbsoluteCaptureTimeExtensionId);
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constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
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const char payload_name[] = "audio";
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const uint8_t payload_type = 127;
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ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
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payload_name, payload_type, 48000, 0, 1500));
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uint8_t payload[] = {47, 11, 32, 93, 89};
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ASSERT_TRUE(rtp_sender_audio_->SendAudio(
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AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
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sizeof(payload), kAbsoluteCaptureTimestampMs));
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auto absolute_capture_time =
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transport_.last_sent_packet()
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.GetExtension<AbsoluteCaptureTimeExtension>();
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EXPECT_TRUE(absolute_capture_time);
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EXPECT_EQ(
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absolute_capture_time->absolute_capture_timestamp,
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Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
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kAbsoluteCaptureTimestampMs)));
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EXPECT_TRUE(
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absolute_capture_time->estimated_capture_clock_offset.has_value());
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EXPECT_EQ(0, *absolute_capture_time->estimated_capture_clock_offset);
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}
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// As RFC4733, named telephone events are carried as part of the audio stream
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// and must use the same sequence number and timestamp base as the regular
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// audio channel.
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// This test checks the marker bit for the first packet and the consequent
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// packets of the same telephone event. Since it is specifically for DTMF
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// events, ignoring audio packets and sending kEmptyFrame instead of those.
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TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
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const char* kDtmfPayloadName = "telephone-event";
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const uint32_t kPayloadFrequency = 8000;
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const uint8_t kPayloadType = 126;
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ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
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kDtmfPayloadName, kPayloadType, kPayloadFrequency, 0, 0));
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// For Telephone events, payload is not added to the registered payload list,
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// it will register only the payload used for audio stream.
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// Registering the payload again for audio stream with different payload name.
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const char* kPayloadName = "payload_name";
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ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
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kPayloadName, kPayloadType, kPayloadFrequency, 1, 0));
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// Start time is arbitrary.
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uint32_t capture_timestamp = fake_clock_.TimeInMilliseconds();
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// DTMF event key=9, duration=500 and attenuationdB=10
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rtp_sender_audio_->SendTelephoneEvent(9, 500, 10);
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// During start, it takes the starting timestamp as last sent timestamp.
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// The duration is calculated as the difference of current and last sent
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// timestamp. So for first call it will skip since the duration is zero.
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ASSERT_TRUE(rtp_sender_audio_->SendAudio(
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AudioFrameType::kEmptyFrame, kPayloadType, capture_timestamp, nullptr, 0,
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/*absolute_capture_time_ms=0*/ 0));
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// DTMF Sample Length is (Frequency/1000) * Duration.
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// So in this case, it is (8000/1000) * 500 = 4000.
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// Sending it as two packets.
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ASSERT_TRUE(rtp_sender_audio_->SendAudio(AudioFrameType::kEmptyFrame,
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kPayloadType,
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capture_timestamp + 2000, nullptr, 0,
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/*absolute_capture_time_ms=0*/ 0));
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// Marker Bit should be set to 1 for first packet.
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EXPECT_TRUE(transport_.last_sent_packet().Marker());
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ASSERT_TRUE(rtp_sender_audio_->SendAudio(AudioFrameType::kEmptyFrame,
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kPayloadType,
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capture_timestamp + 4000, nullptr, 0,
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/*absolute_capture_time_ms=0*/ 0));
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// Marker Bit should be set to 0 for rest of the packets.
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EXPECT_FALSE(transport_.last_sent_packet().Marker());
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}
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} // namespace webrtc
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