TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
302 lines
8.5 KiB
C++
302 lines
8.5 KiB
C++
/*
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* Copyright (c) 2012 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 "voice_engine/voe_base_impl.h"
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#include "common_audio/signal_processing/include/signal_processing_library.h"
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_device/audio_device_impl.h"
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#include "rtc_base/format_macros.h"
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#include "rtc_base/location.h"
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#include "rtc_base/logging.h"
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#include "voice_engine/channel.h"
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#include "voice_engine/include/voe_errors.h"
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#include "voice_engine/voice_engine_impl.h"
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namespace webrtc {
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VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine) {
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if (nullptr == voiceEngine) {
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return nullptr;
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}
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VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
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s->AddRef();
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return s;
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}
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VoEBaseImpl::VoEBaseImpl(voe::SharedData* shared)
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: shared_(shared) {}
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VoEBaseImpl::~VoEBaseImpl() {
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TerminateInternal();
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}
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int VoEBaseImpl::Init(
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AudioDeviceModule* audio_device,
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AudioProcessing* audio_processing,
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const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
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RTC_DCHECK(audio_device);
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rtc::CritScope cs(shared_->crit_sec());
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if (shared_->process_thread()) {
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shared_->process_thread()->Start();
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}
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shared_->set_audio_device(audio_device);
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RTC_DCHECK(decoder_factory);
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decoder_factory_ = decoder_factory;
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return 0;
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}
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void VoEBaseImpl::Terminate() {
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rtc::CritScope cs(shared_->crit_sec());
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TerminateInternal();
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}
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int VoEBaseImpl::CreateChannel() {
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return CreateChannel(ChannelConfig());
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}
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int VoEBaseImpl::CreateChannel(const ChannelConfig& config) {
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rtc::CritScope cs(shared_->crit_sec());
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ChannelConfig config_copy(config);
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config_copy.acm_config.decoder_factory = decoder_factory_;
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voe::ChannelOwner channel_owner =
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shared_->channel_manager().CreateChannel(config_copy);
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return InitializeChannel(&channel_owner);
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}
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int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
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if (channel_owner->channel()->SetEngineInformation(
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*shared_->process_thread(), *shared_->audio_device(),
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shared_->encoder_queue()) != 0) {
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RTC_LOG(LS_ERROR)
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<< "CreateChannel() failed to associate engine and channel."
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" Destroying channel.";
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shared_->channel_manager().DestroyChannel(
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channel_owner->channel()->ChannelId());
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return -1;
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} else if (channel_owner->channel()->Init() != 0) {
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RTC_LOG(LS_ERROR)
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<< "CreateChannel() failed to initialize channel. Destroying"
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" channel.";
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shared_->channel_manager().DestroyChannel(
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channel_owner->channel()->ChannelId());
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return -1;
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}
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return channel_owner->channel()->ChannelId();
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}
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int VoEBaseImpl::DeleteChannel(int channel) {
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rtc::CritScope cs(shared_->crit_sec());
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{
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voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
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voe::Channel* channelPtr = ch.channel();
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if (channelPtr == nullptr) {
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RTC_LOG(LS_ERROR) << "DeleteChannel() failed to locate channel";
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return -1;
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}
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}
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shared_->channel_manager().DestroyChannel(channel);
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if (StopSend() != 0) {
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return -1;
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}
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if (StopPlayout() != 0) {
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return -1;
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}
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return 0;
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}
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int VoEBaseImpl::StartPlayout(int channel) {
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rtc::CritScope cs(shared_->crit_sec());
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voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
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voe::Channel* channelPtr = ch.channel();
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if (channelPtr == nullptr) {
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RTC_LOG(LS_ERROR) << "StartPlayout() failed to locate channel";
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return -1;
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}
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if (channelPtr->Playing()) {
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return 0;
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}
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if (StartPlayout() != 0) {
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RTC_LOG(LS_ERROR) << "StartPlayout() failed to start playout";
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return -1;
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}
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return channelPtr->StartPlayout();
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}
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int VoEBaseImpl::StopPlayout(int channel) {
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rtc::CritScope cs(shared_->crit_sec());
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voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
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voe::Channel* channelPtr = ch.channel();
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if (channelPtr == nullptr) {
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RTC_LOG(LS_ERROR) << "StopPlayout() failed to locate channel";
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return -1;
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}
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if (channelPtr->StopPlayout() != 0) {
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RTC_LOG_F(LS_WARNING) << "StopPlayout() failed to stop playout for channel "
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<< channel;
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}
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return StopPlayout();
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}
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int VoEBaseImpl::StartSend(int channel) {
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rtc::CritScope cs(shared_->crit_sec());
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voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
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voe::Channel* channelPtr = ch.channel();
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if (channelPtr == nullptr) {
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RTC_LOG(LS_ERROR) << "StartSend() failed to locate channel";
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return -1;
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}
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if (channelPtr->Sending()) {
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return 0;
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}
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if (StartSend() != 0) {
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RTC_LOG(LS_ERROR) << "StartSend() failed to start recording";
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return -1;
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}
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return channelPtr->StartSend();
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}
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int VoEBaseImpl::StopSend(int channel) {
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rtc::CritScope cs(shared_->crit_sec());
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voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
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voe::Channel* channelPtr = ch.channel();
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if (channelPtr == nullptr) {
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RTC_LOG(LS_ERROR) << "StopSend() failed to locate channel";
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return -1;
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}
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channelPtr->StopSend();
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return StopSend();
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}
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int32_t VoEBaseImpl::StartPlayout() {
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if (!shared_->audio_device()->Playing()) {
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if (shared_->audio_device()->InitPlayout() != 0) {
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RTC_LOG_F(LS_ERROR) << "Failed to initialize playout";
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return -1;
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}
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if (playout_enabled_ && shared_->audio_device()->StartPlayout() != 0) {
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RTC_LOG_F(LS_ERROR) << "Failed to start playout";
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return -1;
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}
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}
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return 0;
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}
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int32_t VoEBaseImpl::StopPlayout() {
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if (!playout_enabled_) {
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return 0;
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}
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// Stop audio-device playing if no channel is playing out.
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if (shared_->NumOfPlayingChannels() == 0) {
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if (shared_->audio_device()->StopPlayout() != 0) {
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RTC_LOG(LS_ERROR) << "StopPlayout() failed to stop playout";
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return -1;
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}
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}
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return 0;
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}
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int32_t VoEBaseImpl::StartSend() {
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if (!shared_->audio_device()->Recording()) {
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if (shared_->audio_device()->InitRecording() != 0) {
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RTC_LOG_F(LS_ERROR) << "Failed to initialize recording";
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return -1;
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}
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if (recording_enabled_ && shared_->audio_device()->StartRecording() != 0) {
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RTC_LOG_F(LS_ERROR) << "Failed to start recording";
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return -1;
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}
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}
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return 0;
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}
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int32_t VoEBaseImpl::StopSend() {
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if (!recording_enabled_) {
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return 0;
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}
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// Stop audio-device recording if no channel is recording.
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if (shared_->NumOfSendingChannels() == 0) {
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if (shared_->audio_device()->StopRecording() != 0) {
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RTC_LOG(LS_ERROR) << "StopSend() failed to stop recording";
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return -1;
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}
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}
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return 0;
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}
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int32_t VoEBaseImpl::SetPlayout(bool enabled) {
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RTC_LOG(INFO) << "SetPlayout(" << enabled << ")";
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if (playout_enabled_ == enabled) {
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return 0;
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}
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playout_enabled_ = enabled;
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if (shared_->NumOfPlayingChannels() == 0) {
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// If there are no channels attempting to play out yet, there's nothing to
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// be done; we should be in a "not playing out" state either way.
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return 0;
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}
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int32_t ret;
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if (enabled) {
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ret = shared_->audio_device()->StartPlayout();
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if (ret != 0) {
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RTC_LOG(LS_ERROR) << "SetPlayout(true) failed to start playout";
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}
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} else {
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ret = shared_->audio_device()->StopPlayout();
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if (ret != 0) {
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RTC_LOG(LS_ERROR) << "SetPlayout(false) failed to stop playout";
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}
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}
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return ret;
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}
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int32_t VoEBaseImpl::SetRecording(bool enabled) {
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RTC_LOG(INFO) << "SetRecording(" << enabled << ")";
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if (recording_enabled_ == enabled) {
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return 0;
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}
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recording_enabled_ = enabled;
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if (shared_->NumOfSendingChannels() == 0) {
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// If there are no channels attempting to record out yet, there's nothing to
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// be done; we should be in a "not recording" state either way.
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return 0;
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}
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int32_t ret;
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if (enabled) {
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ret = shared_->audio_device()->StartRecording();
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if (ret != 0) {
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RTC_LOG(LS_ERROR) << "SetRecording(true) failed to start recording";
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}
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} else {
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ret = shared_->audio_device()->StopRecording();
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if (ret != 0) {
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RTC_LOG(LS_ERROR) << "SetRecording(false) failed to stop recording";
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}
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}
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return ret;
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}
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void VoEBaseImpl::TerminateInternal() {
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// Delete any remaining channel objects
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shared_->channel_manager().DestroyAllChannels();
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if (shared_->process_thread()) {
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shared_->process_thread()->Stop();
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}
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shared_->set_audio_device(nullptr);
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}
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} // namespace webrtc
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