webrtc_m130/media/engine/apm_helpers.cc
Fredrik Solenberg 2a8779763a Remove voe::TransmitMixer
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}
2017-12-15 16:48:57 +00:00

171 lines
5.7 KiB
C++

/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "media/engine/apm_helpers.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace apm_helpers {
void Init(AudioProcessing* apm) {
RTC_DCHECK(apm);
constexpr int kMinVolumeLevel = 0;
constexpr int kMaxVolumeLevel = 255;
// This is the initialization which used to happen in VoEBase::Init(), but
// which is not covered by the WVoE::ApplyOptions().
if (apm->echo_cancellation()->enable_drift_compensation(false) != 0) {
RTC_DLOG(LS_ERROR) << "Failed to disable drift compensation.";
}
GainControl* gc = apm->gain_control();
if (gc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
RTC_DLOG(LS_ERROR) << "Failed to set analog level limits with minimum: "
<< kMinVolumeLevel << " and maximum: " << kMaxVolumeLevel;
}
}
AgcConfig GetAgcConfig(AudioProcessing* apm) {
RTC_DCHECK(apm);
AgcConfig result;
result.targetLeveldBOv = apm->gain_control()->target_level_dbfs();
result.digitalCompressionGaindB = apm->gain_control()->compression_gain_db();
result.limiterEnable = apm->gain_control()->is_limiter_enabled();
return result;
}
void SetAgcConfig(AudioProcessing* apm,
const AgcConfig& config) {
RTC_DCHECK(apm);
GainControl* gc = apm->gain_control();
if (gc->set_target_level_dbfs(config.targetLeveldBOv) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set target level: "
<< config.targetLeveldBOv;
}
if (gc->set_compression_gain_db(config.digitalCompressionGaindB) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set compression gain: "
<< config.digitalCompressionGaindB;
}
if (gc->enable_limiter(config.limiterEnable) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set limiter on/off: "
<< config.limiterEnable;
}
}
void SetAgcStatus(AudioProcessing* apm,
bool enable) {
RTC_DCHECK(apm);
#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
GainControl::Mode agc_mode = GainControl::kFixedDigital;
#else
GainControl::Mode agc_mode = GainControl::kAdaptiveAnalog;
#endif
GainControl* gc = apm->gain_control();
if (gc->set_mode(agc_mode) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set AGC mode: " << agc_mode;
return;
}
if (gc->Enable(enable) != 0) {
RTC_LOG(LS_ERROR) << "Failed to enable/disable AGC: " << enable;
return;
}
RTC_LOG(LS_INFO) << "AGC set to " << enable << " with mode " << agc_mode;
}
void SetEcStatus(AudioProcessing* apm,
bool enable,
EcModes mode) {
RTC_DCHECK(apm);
RTC_DCHECK(mode == kEcConference || mode == kEcAecm) << "mode: " << mode;
EchoCancellation* ec = apm->echo_cancellation();
EchoControlMobile* ecm = apm->echo_control_mobile();
if (mode == kEcConference) {
// Disable the AECM before enabling the AEC.
if (enable && ecm->is_enabled() && ecm->Enable(false) != 0) {
RTC_LOG(LS_ERROR) << "Failed to disable AECM.";
return;
}
if (ec->Enable(enable) != 0) {
RTC_LOG(LS_ERROR) << "Failed to enable/disable AEC: " << enable;
return;
}
if (ec->set_suppression_level(EchoCancellation::kHighSuppression)
!= 0) {
RTC_LOG(LS_ERROR) << "Failed to set high AEC aggressiveness.";
return;
}
} else {
// Disable the AEC before enabling the AECM.
if (enable && ec->is_enabled() && ec->Enable(false) != 0) {
RTC_LOG(LS_ERROR) << "Failed to disable AEC.";
return;
}
if (ecm->Enable(enable) != 0) {
RTC_LOG(LS_ERROR) << "Failed to enable/disable AECM: " << enable;
return;
}
}
RTC_LOG(LS_INFO) << "Echo control set to " << enable << " with mode " << mode;
}
void SetEcMetricsStatus(AudioProcessing* apm, bool enable) {
RTC_DCHECK(apm);
if ((apm->echo_cancellation()->enable_metrics(enable) != 0) ||
(apm->echo_cancellation()->enable_delay_logging(enable) != 0)) {
RTC_LOG(LS_ERROR) << "Failed to enable/disable EC metrics: " << enable;
return;
}
RTC_LOG(LS_INFO) << "EC metrics set to " << enable;
}
void SetAecmMode(AudioProcessing* apm, bool enable) {
RTC_DCHECK(apm);
EchoControlMobile* ecm = apm->echo_control_mobile();
RTC_DCHECK_EQ(EchoControlMobile::kSpeakerphone, ecm->routing_mode());
if (ecm->enable_comfort_noise(enable) != 0) {
RTC_LOG(LS_ERROR) << "Failed to enable/disable CNG: " << enable;
return;
}
RTC_LOG(LS_INFO) << "CNG set to " << enable;
}
void SetNsStatus(AudioProcessing* apm, bool enable) {
RTC_DCHECK(apm);
NoiseSuppression* ns = apm->noise_suppression();
if (ns->set_level(NoiseSuppression::kHigh) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set high NS level.";
return;
}
if (ns->Enable(enable) != 0) {
RTC_LOG(LS_ERROR) << "Failed to enable/disable NS: " << enable;
return;
}
RTC_LOG(LS_INFO) << "NS set to " << enable;
}
void SetTypingDetectionStatus(AudioProcessing* apm, bool enable) {
RTC_DCHECK(apm);
VoiceDetection* vd = apm->voice_detection();
if (vd->Enable(enable)) {
RTC_LOG(LS_ERROR) << "Failed to enable/disable VAD: " << enable;
return;
}
if (vd->set_likelihood(VoiceDetection::kVeryLowLikelihood)) {
RTC_LOG(LS_ERROR) << "Failed to set low VAD likelihood.";
return;
}
RTC_LOG(LS_INFO) << "VAD set to " << enable << " for typing detection.";
}
} // namespace apm_helpers
} // namespace webrtc