From dcb09ff21808894c4866073b17d49a9048d69340 Mon Sep 17 00:00:00 2001 From: Jakob Ivarsson Date: Wed, 25 Jan 2023 20:03:56 +0100 Subject: [PATCH] Reset encoder when audio send stream is stopped. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This is to clear any remaining buffers and other state such as the next packet timestamp. Bug: webrtc:12397 Change-Id: I2ef9a6f7254d82a69a2896ec8aa619ced2694fb8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291327 Reviewed-by: Henrik Lundin Commit-Queue: Jakob Ivarsson‎ Cr-Commit-Position: refs/heads/main@{#39206} --- audio/BUILD.gn | 4 ++ audio/channel_send.cc | 4 +- audio/channel_send_unittest.cc | 113 +++++++++++++++++++++++++++++++++ 3 files changed, 120 insertions(+), 1 deletion(-) create mode 100644 audio/channel_send_unittest.cc diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 919140e15c..6ca47d4d1e 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -151,6 +151,7 @@ if (rtc_include_tests) { "audio_state_unittest.cc", "channel_receive_frame_transformer_delegate_unittest.cc", "channel_send_frame_transformer_delegate_unittest.cc", + "channel_send_unittest.cc", "mock_voe_channel_proxy.h", "remix_resample_unittest.cc", "test/audio_stats_test.cc", @@ -165,8 +166,10 @@ if (rtc_include_tests) { "../api:mock_audio_mixer", "../api:mock_frame_decryptor", "../api:mock_frame_encryptor", + "../api:scoped_refptr", "../api/audio:audio_frame_api", "../api/audio_codecs:audio_codecs_api", + "../api/audio_codecs:builtin_audio_encoder_factory", "../api/audio_codecs/opus:audio_decoder_opus", "../api/audio_codecs/opus:audio_encoder_opus", "../api/crypto:frame_decryptor_interface", @@ -174,6 +177,7 @@ if (rtc_include_tests) { "../api/task_queue:default_task_queue_factory", "../api/task_queue/test:mock_task_queue_base", "../api/units:time_delta", + "../api/units:timestamp", "../call:mock_bitrate_allocator", "../call:mock_call_interfaces", "../call:mock_rtp_interfaces", diff --git a/audio/channel_send.cc b/audio/channel_send.cc index 361880d68f..5b0858a76a 100644 --- a/audio/channel_send.cc +++ b/audio/channel_send.cc @@ -540,10 +540,12 @@ void ChannelSend::StopSend() { sending_ = false; encoder_queue_is_active_.store(false); - // Wait until all pending encode tasks are executed. + // Wait until all pending encode tasks are executed and clear any remaining + // buffers in the encoder. rtc::Event flush; encoder_queue_.PostTask([this, &flush]() { RTC_DCHECK_RUN_ON(&encoder_queue_); + CallEncoder([](AudioEncoder* encoder) { encoder->Reset(); }); flush.Set(); }); flush.Wait(rtc::Event::kForever); diff --git a/audio/channel_send_unittest.cc b/audio/channel_send_unittest.cc new file mode 100644 index 0000000000..50d8368d4a --- /dev/null +++ b/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 + +#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 CreateAudioFrame() { + auto frame = std::make_unique(); + 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 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 channel = CreateChannelSend(); + rtc::scoped_refptr encoder_factory = + CreateBuiltinAudioEncoderFactory(); + std::unique_ptr 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