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functionalities doing sample-rate conversions: -The implicit resampling done in the AudioBuffer CopyTo, CopyFrom, InterleaveTo and DeinterleaveFrom methods. -The multi-band splitting scheme. The selection of rates in these have been difficult and complicated, partly due to that the APM API which allows for activating the APM submodules without notifying the APM. This CL adds functionality that for each capture frame polls all submodules for whether they are active or not and compares this against a cached result. Furthermore, new functionality is added that based on the results of the comparison do a reinitialization of the APM. This has several advantages -The code deciding on whether to analysis and synthesis is needed for the bandsplitting can be much simplified and centralized. -The selection of the processing rate can be done such as to avoid the implicit resampling that was in some cases unnecessarily done. -The optimization for whether an output copy is needed that was done to improve performance due to the implicit resampling is no longer needed, which simplifies the code and makes it less error-prone in the sense that is no longer neccessary to keep track of whether any module has changed the signal. Finally, it should be noted that the polling of the state for all the submodules was done previously as well, but in a less obvious and distributed manner. BUG=webrtc:6181, webrtc:6220, webrtc:5298, webrtc:6296, webrtc:6298, webrtc:6297 Review-Url: https://codereview.webrtc.org/2304123002 Cr-Commit-Position: refs/heads/master@{#14175}
94 lines
3.1 KiB
C++
94 lines
3.1 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 WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
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#include <memory>
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#include <vector>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/swap_queue.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
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namespace webrtc {
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class AudioBuffer;
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class EchoControlMobileImpl : public EchoControlMobile {
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public:
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EchoControlMobileImpl(rtc::CriticalSection* crit_render,
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rtc::CriticalSection* crit_capture);
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~EchoControlMobileImpl() override;
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int ProcessRenderAudio(const AudioBuffer* audio);
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int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
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// EchoControlMobile implementation.
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bool is_enabled() const override;
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RoutingMode routing_mode() const override;
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bool is_comfort_noise_enabled() const override;
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void Initialize(int sample_rate_hz,
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size_t num_reverse_channels,
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size_t num_output_channels);
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// Reads render side data that has been queued on the render call.
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void ReadQueuedRenderData();
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private:
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class Canceller;
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struct StreamProperties;
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// EchoControlMobile implementation.
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int Enable(bool enable) override;
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int set_routing_mode(RoutingMode mode) override;
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int enable_comfort_noise(bool enable) override;
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int SetEchoPath(const void* echo_path, size_t size_bytes) override;
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int GetEchoPath(void* echo_path, size_t size_bytes) const override;
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size_t num_handles_required() const;
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void AllocateRenderQueue();
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int Configure();
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rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
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rtc::CriticalSection* const crit_capture_;
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bool enabled_ = false;
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RoutingMode routing_mode_ GUARDED_BY(crit_capture_);
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bool comfort_noise_enabled_ GUARDED_BY(crit_capture_);
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unsigned char* external_echo_path_ GUARDED_BY(crit_render_)
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GUARDED_BY(crit_capture_);
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size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
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GUARDED_BY(crit_capture_);
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std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
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std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
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// Lock protection not needed.
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std::unique_ptr<
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SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
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render_signal_queue_;
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std::vector<std::unique_ptr<Canceller>> cancellers_;
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std::unique_ptr<StreamProperties> stream_properties_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoControlMobileImpl);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
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