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}diff --git a/audio/audio_state.cc b/audio/audio_state.cc
index 5a30c53..a83b681 100644
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@@ -10,13 +10,16 @@
#include "audio/audio_state.h"
+#include <algorithm>
+#include <utility>
+#include <vector>
+
#include "modules/audio_device/include/audio_device.h"
#include "rtc_base/atomicops.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/thread.h"
-#include "voice_engine/transmit_mixer.h"
namespace webrtc {
namespace internal {
@@ -24,15 +27,16 @@
AudioState::AudioState(const AudioState::Config& config)
: config_(config),
voe_base_(config.voice_engine),
- audio_transport_proxy_(voe_base_->audio_transport(),
- config_.audio_processing.get(),
- config_.audio_mixer) {
+ audio_transport_(config_.audio_mixer,
+ config_.audio_processing.get(),
+ config_.audio_device_module.get()) {
process_thread_checker_.DetachFromThread();
RTC_DCHECK(config_.audio_mixer);
}
AudioState::~AudioState() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(sending_streams_.empty());
}
VoiceEngine* AudioState::voice_engine() {
@@ -47,11 +51,23 @@
bool AudioState::typing_noise_detected() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- // TODO(solenberg): Remove const_cast once AudioState owns transmit mixer
- // functionality.
- voe::TransmitMixer* transmit_mixer =
- const_cast<AudioState*>(this)->voe_base_->transmit_mixer();
- return transmit_mixer->typing_noise_detected();
+ return audio_transport_.typing_noise_detected();
+}
+
+void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
+ int sample_rate_hz, size_t num_channels) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ auto& properties = sending_streams_[stream];
+ properties.sample_rate_hz = sample_rate_hz;
+ properties.num_channels = num_channels;
+ UpdateAudioTransportWithSendingStreams();
+}
+
+void AudioState::RemoveSendingStream(webrtc::AudioSendStream* stream) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ auto count = sending_streams_.erase(stream);
+ RTC_DCHECK_EQ(1, count);
+ UpdateAudioTransportWithSendingStreams();
}
void AudioState::SetPlayout(bool enabled) {
@@ -61,33 +77,47 @@
if (enabled == currently_enabled) {
return;
}
- VoEBase* const voe = VoEBase::GetInterface(voice_engine());
- RTC_DCHECK(voe);
if (enabled) {
null_audio_poller_.reset();
}
// Will stop/start playout of the underlying device, if necessary, and
// remember the setting for when it receives subsequent calls of
// StartPlayout.
- voe->SetPlayout(enabled);
+ voe_base_->SetPlayout(enabled);
if (!enabled) {
null_audio_poller_ =
- rtc::MakeUnique<NullAudioPoller>(&audio_transport_proxy_);
+ rtc::MakeUnique<NullAudioPoller>(&audio_transport_);
}
- voe->Release();
}
void AudioState::SetRecording(bool enabled) {
RTC_LOG(INFO) << "SetRecording(" << enabled << ")";
RTC_DCHECK(thread_checker_.CalledOnValidThread());
// TODO(henrika): keep track of state as in SetPlayout().
- VoEBase* const voe = VoEBase::GetInterface(voice_engine());
- RTC_DCHECK(voe);
// Will stop/start recording of the underlying device, if necessary, and
// remember the setting for when it receives subsequent calls of
// StartPlayout.
- voe->SetRecording(enabled);
- voe->Release();
+ voe_base_->SetRecording(enabled);
+}
+
+AudioState::Stats AudioState::GetAudioInputStats() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ const voe::AudioLevel& audio_level = audio_transport_.audio_level();
+ Stats result;
+ result.audio_level = audio_level.LevelFullRange();
+ RTC_DCHECK_LE(0, result.audio_level);
+ RTC_DCHECK_GE(32767, result.audio_level);
+ result.quantized_audio_level = audio_level.Level();
+ RTC_DCHECK_LE(0, result.quantized_audio_level);
+ RTC_DCHECK_GE(9, result.quantized_audio_level);
+ result.total_energy = audio_level.TotalEnergy();
+ result.total_duration = audio_level.TotalDuration();
+ return result;
+}
+
+void AudioState::SetStereoChannelSwapping(bool enable) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ audio_transport_.SetStereoChannelSwapping(enable);
}
// Reference count; implementation copied from rtc::RefCountedObject.
@@ -103,6 +133,20 @@
}
return rtc::RefCountReleaseStatus::kOtherRefsRemained;
}
+
+void AudioState::UpdateAudioTransportWithSendingStreams() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ std::vector<AudioSendStream*> sending_streams;
+ int max_sample_rate_hz = 8000;
+ size_t max_num_channels = 1;
+ for (const auto& kv : sending_streams_) {
+ sending_streams.push_back(kv.first);
+ max_sample_rate_hz = std::max(max_sample_rate_hz, kv.second.sample_rate_hz);
+ max_num_channels = std::max(max_num_channels, kv.second.num_channels);
+ }
+ audio_transport_.UpdateSendingStreams(std::move(sending_streams),
+ max_sample_rate_hz, max_num_channels);
+}
} // namespace internal
rtc::scoped_refptr<AudioState> AudioState::Create(
