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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}
85 lines
3.1 KiB
C++
85 lines
3.1 KiB
C++
/*
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* Copyright (c) 2015 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 CALL_AUDIO_STATE_H_
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#define CALL_AUDIO_STATE_H_
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#include "api/audio/audio_mixer.h"
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#include "rtc_base/refcount.h"
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#include "rtc_base/scoped_ref_ptr.h"
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namespace webrtc {
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class AudioDeviceModule;
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class AudioProcessing;
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class AudioTransport;
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class VoiceEngine;
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// WORK IN PROGRESS
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// This class is under development and is not yet intended for for use outside
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// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
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// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
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// AudioState holds the state which must be shared between multiple instances of
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// webrtc::Call for audio processing purposes.
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class AudioState : public rtc::RefCountInterface {
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public:
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struct Config {
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// VoiceEngine used for audio streams and audio/video synchronization.
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// AudioState will tickle the VoE refcount to keep it alive for as long as
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// the AudioState itself.
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VoiceEngine* voice_engine = nullptr;
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// The audio mixer connected to active receive streams. One per
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// AudioState.
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rtc::scoped_refptr<AudioMixer> audio_mixer;
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// The audio processing module.
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rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
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// TODO(solenberg): Temporary: audio device module.
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rtc::scoped_refptr<webrtc::AudioDeviceModule> audio_device_module;
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};
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struct Stats {
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// Audio peak level (max(abs())), linearly on the interval [0,32767].
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int32_t audio_level = -1;
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// Audio peak level (max(abs())), logarithmically on the interval [0,9].
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int8_t quantized_audio_level = -1;
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// See: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
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double total_energy = 0.0f;
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double total_duration = 0.0f;
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};
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virtual AudioProcessing* audio_processing() = 0;
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virtual AudioTransport* audio_transport() = 0;
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// Enable/disable playout of the audio channels. Enabled by default.
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// This will stop playout of the underlying audio device but start a task
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// which will poll for audio data every 10ms to ensure that audio processing
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// happens and the audio stats are updated.
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virtual void SetPlayout(bool enabled) = 0;
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// Enable/disable recording of the audio channels. Enabled by default.
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// This will stop recording of the underlying audio device and no audio
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// packets will be encoded or transmitted.
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virtual void SetRecording(bool enabled) = 0;
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virtual Stats GetAudioInputStats() const = 0;
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virtual void SetStereoChannelSwapping(bool enable) = 0;
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// TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
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static rtc::scoped_refptr<AudioState> Create(
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const AudioState::Config& config);
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virtual ~AudioState() {}
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};
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} // namespace webrtc
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#endif // CALL_AUDIO_STATE_H_
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