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8f5787a919
* VoEBase contains only stub methods (until downstream code is updated). * voe::Channel and ChannelProxy classes remain, but are now created internally to the streams. As a result, internal::Audio[Receive|Send]Stream can have a ChannelProxy injected for testing. * Stream classes share Call::module_process_thread_ for their RtpRtcp modules, rather than using a separate thread shared only among audio streams. * voe::Channel instances use Call::worker_queue_ for encoding packets, rather than having a separate queue for audio (send) streams. Bug: webrtc:4690 Change-Id: I8059ef224ad13aa0a6ded2cafc52599c7f64d68d Reviewed-on: https://webrtc-review.googlesource.com/34640 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21578}
78 lines
2.7 KiB
C++
78 lines
2.7 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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// 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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// TODO(solenberg): Remove once clients don't use it anymore.
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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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