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}
101 lines
3.8 KiB
C++
101 lines
3.8 KiB
C++
/*
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* Copyright (c) 2016 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 AUDIO_AUDIO_TRANSPORT_IMPL_H_
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#define AUDIO_AUDIO_TRANSPORT_IMPL_H_
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#include <vector>
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#include "api/audio/audio_mixer.h"
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#include "common_audio/resampler/include/push_resampler.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/typing_detection.h"
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#include "rtc_base/constructormagic.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/thread_annotations.h"
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#include "voice_engine/audio_level.h"
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namespace webrtc {
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class AudioSendStream;
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class AudioTransportImpl : public AudioTransport {
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public:
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AudioTransportImpl(AudioMixer* mixer,
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AudioProcessing* audio_processing,
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AudioDeviceModule* audio_device_module);
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~AudioTransportImpl() override;
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int32_t RecordedDataIsAvailable(const void* audioSamples,
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const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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const uint32_t totalDelayMS,
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const int32_t clockDrift,
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const uint32_t currentMicLevel,
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const bool keyPressed,
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uint32_t& newMicLevel) override;
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int32_t NeedMorePlayData(const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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void* audioSamples,
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size_t& nSamplesOut,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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void PullRenderData(int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames,
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void* audio_data,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) override;
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void UpdateSendingStreams(std::vector<AudioSendStream*> streams,
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int send_sample_rate_hz, size_t send_num_channels);
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void SetStereoChannelSwapping(bool enable);
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bool typing_noise_detected() const;
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const voe::AudioLevel& audio_level() const {
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return audio_level_;
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}
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private:
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// Shared.
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AudioProcessing* audio_processing_ = nullptr;
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// Capture side.
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rtc::CriticalSection capture_lock_;
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std::vector<AudioSendStream*> sending_streams_ RTC_GUARDED_BY(capture_lock_);
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int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000;
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size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
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bool typing_noise_detected_ RTC_GUARDED_BY(capture_lock_) = false;
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bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
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AudioDeviceModule* audio_device_module_ = nullptr;
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PushResampler<int16_t> capture_resampler_;
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voe::AudioLevel audio_level_;
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TypingDetection typing_detection_;
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// Render side.
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rtc::scoped_refptr<AudioMixer> mixer_;
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AudioFrame mixed_frame_;
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// Converts mixed audio to the audio device output rate.
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PushResampler<int16_t> render_resampler_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportImpl);
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};
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} // namespace webrtc
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#endif // AUDIO_AUDIO_TRANSPORT_IMPL_H_
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