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}
64 lines
2.0 KiB
C++
64 lines
2.0 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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#ifndef VOICE_ENGINE_SHARED_DATA_H_
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#define VOICE_ENGINE_SHARED_DATA_H_
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#include <memory>
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#include "modules/audio_device/include/audio_device.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/scoped_ref_ptr.h"
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#include "rtc_base/task_queue.h"
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#include "rtc_base/thread_annotations.h"
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#include "rtc_base/thread_checker.h"
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#include "voice_engine/channel_manager.h"
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class ProcessThread;
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namespace webrtc {
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namespace voe {
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class SharedData
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{
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public:
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// Public accessors.
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uint32_t instance_id() const { return _instanceId; }
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ChannelManager& channel_manager() { return _channelManager; }
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AudioDeviceModule* audio_device() { return _audioDevicePtr.get(); }
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void set_audio_device(
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const rtc::scoped_refptr<AudioDeviceModule>& audio_device);
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rtc::CriticalSection* crit_sec() { return &_apiCritPtr; }
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ProcessThread* process_thread() { return _moduleProcessThreadPtr.get(); }
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rtc::TaskQueue* encoder_queue();
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int NumOfSendingChannels();
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int NumOfPlayingChannels();
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protected:
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rtc::ThreadChecker construction_thread_;
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const uint32_t _instanceId;
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rtc::CriticalSection _apiCritPtr;
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ChannelManager _channelManager;
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rtc::scoped_refptr<AudioDeviceModule> _audioDevicePtr;
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std::unique_ptr<ProcessThread> _moduleProcessThreadPtr;
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// |encoder_queue| is defined last to ensure all pending tasks are cancelled
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// and deleted before any other members.
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rtc::TaskQueue encoder_queue_ RTC_ACCESS_ON(construction_thread_);
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SharedData();
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virtual ~SharedData();
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};
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} // namespace voe
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} // namespace webrtc
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#endif // VOICE_ENGINE_SHARED_DATA_H_
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