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}
171 lines
5.7 KiB
C++
171 lines
5.7 KiB
C++
/*
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* Copyright (c) 2017 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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#include "media/engine/apm_helpers.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 "rtc_base/logging.h"
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namespace webrtc {
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namespace apm_helpers {
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void Init(AudioProcessing* apm) {
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RTC_DCHECK(apm);
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constexpr int kMinVolumeLevel = 0;
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constexpr int kMaxVolumeLevel = 255;
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// This is the initialization which used to happen in VoEBase::Init(), but
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// which is not covered by the WVoE::ApplyOptions().
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if (apm->echo_cancellation()->enable_drift_compensation(false) != 0) {
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RTC_DLOG(LS_ERROR) << "Failed to disable drift compensation.";
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}
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GainControl* gc = apm->gain_control();
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if (gc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
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RTC_DLOG(LS_ERROR) << "Failed to set analog level limits with minimum: "
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<< kMinVolumeLevel << " and maximum: " << kMaxVolumeLevel;
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}
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}
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AgcConfig GetAgcConfig(AudioProcessing* apm) {
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RTC_DCHECK(apm);
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AgcConfig result;
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result.targetLeveldBOv = apm->gain_control()->target_level_dbfs();
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result.digitalCompressionGaindB = apm->gain_control()->compression_gain_db();
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result.limiterEnable = apm->gain_control()->is_limiter_enabled();
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return result;
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}
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void SetAgcConfig(AudioProcessing* apm,
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const AgcConfig& config) {
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RTC_DCHECK(apm);
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GainControl* gc = apm->gain_control();
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if (gc->set_target_level_dbfs(config.targetLeveldBOv) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set target level: "
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<< config.targetLeveldBOv;
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}
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if (gc->set_compression_gain_db(config.digitalCompressionGaindB) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set compression gain: "
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<< config.digitalCompressionGaindB;
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}
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if (gc->enable_limiter(config.limiterEnable) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set limiter on/off: "
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<< config.limiterEnable;
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}
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}
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void SetAgcStatus(AudioProcessing* apm,
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bool enable) {
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RTC_DCHECK(apm);
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#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
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GainControl::Mode agc_mode = GainControl::kFixedDigital;
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#else
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GainControl::Mode agc_mode = GainControl::kAdaptiveAnalog;
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#endif
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GainControl* gc = apm->gain_control();
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if (gc->set_mode(agc_mode) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set AGC mode: " << agc_mode;
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return;
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}
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if (gc->Enable(enable) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to enable/disable AGC: " << enable;
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return;
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}
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RTC_LOG(LS_INFO) << "AGC set to " << enable << " with mode " << agc_mode;
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}
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void SetEcStatus(AudioProcessing* apm,
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bool enable,
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EcModes mode) {
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RTC_DCHECK(apm);
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RTC_DCHECK(mode == kEcConference || mode == kEcAecm) << "mode: " << mode;
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EchoCancellation* ec = apm->echo_cancellation();
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EchoControlMobile* ecm = apm->echo_control_mobile();
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if (mode == kEcConference) {
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// Disable the AECM before enabling the AEC.
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if (enable && ecm->is_enabled() && ecm->Enable(false) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to disable AECM.";
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return;
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}
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if (ec->Enable(enable) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to enable/disable AEC: " << enable;
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return;
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}
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if (ec->set_suppression_level(EchoCancellation::kHighSuppression)
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!= 0) {
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RTC_LOG(LS_ERROR) << "Failed to set high AEC aggressiveness.";
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return;
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}
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} else {
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// Disable the AEC before enabling the AECM.
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if (enable && ec->is_enabled() && ec->Enable(false) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to disable AEC.";
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return;
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}
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if (ecm->Enable(enable) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to enable/disable AECM: " << enable;
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return;
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}
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}
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RTC_LOG(LS_INFO) << "Echo control set to " << enable << " with mode " << mode;
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}
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void SetEcMetricsStatus(AudioProcessing* apm, bool enable) {
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RTC_DCHECK(apm);
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if ((apm->echo_cancellation()->enable_metrics(enable) != 0) ||
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(apm->echo_cancellation()->enable_delay_logging(enable) != 0)) {
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RTC_LOG(LS_ERROR) << "Failed to enable/disable EC metrics: " << enable;
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return;
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}
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RTC_LOG(LS_INFO) << "EC metrics set to " << enable;
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}
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void SetAecmMode(AudioProcessing* apm, bool enable) {
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RTC_DCHECK(apm);
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EchoControlMobile* ecm = apm->echo_control_mobile();
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RTC_DCHECK_EQ(EchoControlMobile::kSpeakerphone, ecm->routing_mode());
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if (ecm->enable_comfort_noise(enable) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to enable/disable CNG: " << enable;
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return;
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}
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RTC_LOG(LS_INFO) << "CNG set to " << enable;
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}
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void SetNsStatus(AudioProcessing* apm, bool enable) {
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RTC_DCHECK(apm);
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NoiseSuppression* ns = apm->noise_suppression();
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if (ns->set_level(NoiseSuppression::kHigh) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set high NS level.";
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return;
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}
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if (ns->Enable(enable) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to enable/disable NS: " << enable;
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return;
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}
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RTC_LOG(LS_INFO) << "NS set to " << enable;
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}
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void SetTypingDetectionStatus(AudioProcessing* apm, bool enable) {
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RTC_DCHECK(apm);
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VoiceDetection* vd = apm->voice_detection();
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if (vd->Enable(enable)) {
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RTC_LOG(LS_ERROR) << "Failed to enable/disable VAD: " << enable;
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return;
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}
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if (vd->set_likelihood(VoiceDetection::kVeryLowLikelihood)) {
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RTC_LOG(LS_ERROR) << "Failed to set low VAD likelihood.";
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return;
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}
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RTC_LOG(LS_INFO) << "VAD set to " << enable << " for typing detection.";
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}
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} // namespace apm_helpers
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} // namespace webrtc
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