2a8779763a
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}
76 lines
1.8 KiB
C++
76 lines
1.8 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/shared_data.h"
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#include "voice_engine/channel.h"
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namespace webrtc {
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namespace voe {
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static int32_t _gInstanceCounter = 0;
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SharedData::SharedData()
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: _instanceId(++_gInstanceCounter),
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_channelManager(_gInstanceCounter),
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_audioDevicePtr(NULL),
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_moduleProcessThreadPtr(ProcessThread::Create("VoiceProcessThread")),
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encoder_queue_("AudioEncoderQueue") {
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}
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SharedData::~SharedData()
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{
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if (_audioDevicePtr) {
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_audioDevicePtr->Release();
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}
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_moduleProcessThreadPtr->Stop();
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}
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rtc::TaskQueue* SharedData::encoder_queue() {
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RTC_DCHECK_RUN_ON(&construction_thread_);
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return &encoder_queue_;
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}
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void SharedData::set_audio_device(
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const rtc::scoped_refptr<AudioDeviceModule>& audio_device) {
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_audioDevicePtr = audio_device;
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}
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int SharedData::NumOfSendingChannels() {
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ChannelManager::Iterator it(&_channelManager);
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int sending_channels = 0;
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for (ChannelManager::Iterator it(&_channelManager); it.IsValid();
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it.Increment()) {
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if (it.GetChannel()->Sending())
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++sending_channels;
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}
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return sending_channels;
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}
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int SharedData::NumOfPlayingChannels() {
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ChannelManager::Iterator it(&_channelManager);
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int playout_channels = 0;
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for (ChannelManager::Iterator it(&_channelManager); it.IsValid();
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it.Increment()) {
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if (it.GetChannel()->Playing())
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++playout_channels;
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}
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return playout_channels;
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}
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} // namespace voe
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} // namespace webrtc
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