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/*
* Copyright (c) 2015 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 "audio/audio_state.h"
#include <algorithm>
#include <memory>
#include <utility>
#include <vector>
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
#include "modules/audio_device/include/audio_device.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace internal {
AudioState::AudioState(const AudioState::Config& config)
: config_(config),
audio_transport_(config_.audio_mixer.get(),
config_.audio_processing.get(),
config_.async_audio_processing_factory.get()) {
RTC_DCHECK(config_.audio_mixer);
RTC_DCHECK(config_.audio_device_module);
}
AudioState::~AudioState() {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK(receiving_streams_.empty());
RTC_DCHECK(sending_streams_.empty());
RTC_DCHECK(!null_audio_poller_.Running());
}
AudioProcessing* AudioState::audio_processing() {
return config_.audio_processing.get();
}
AudioTransport* AudioState::audio_transport() {
return &audio_transport_;
}
void AudioState::AddReceivingStream(
webrtc::AudioReceiveStreamInterface* stream) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_DCHECK_EQ(0, receiving_streams_.count(stream));
receiving_streams_.insert(stream);
if (!config_.audio_mixer->AddSource(
static_cast<AudioReceiveStreamImpl*>(stream))) {
RTC_DLOG(LS_ERROR) << "Failed to add source to mixer.";
}
// Make sure playback is initialized; start playing if enabled.
UpdateNullAudioPollerState();
auto* adm = config_.audio_device_module.get();
if (!adm->Playing()) {
if (adm->InitPlayout() == 0) {
if (playout_enabled_) {
adm->StartPlayout();
}
} else {
RTC_DLOG_F(LS_ERROR) << "Failed to initialize playout.";
}
}
}
void AudioState::RemoveReceivingStream(
webrtc::AudioReceiveStreamInterface* stream) {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto count = receiving_streams_.erase(stream);
RTC_DCHECK_EQ(1, count);
config_.audio_mixer->RemoveSource(
static_cast<AudioReceiveStreamImpl*>(stream));
UpdateNullAudioPollerState();
if (receiving_streams_.empty()) {
config_.audio_device_module->StopPlayout();
}
}
void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
int sample_rate_hz,
size_t num_channels) {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto& properties = sending_streams_[stream];
properties.sample_rate_hz = sample_rate_hz;
properties.num_channels = num_channels;
UpdateAudioTransportWithSendingStreams();
// Make sure recording is initialized; start recording if enabled.
auto* adm = config_.audio_device_module.get();
if (!adm->Recording()) {
if (adm->InitRecording() == 0) {
if (recording_enabled_) {
adm->StartRecording();
}
} else {
RTC_DLOG_F(LS_ERROR) << "Failed to initialize recording.";
}
}
}
void AudioState::RemoveSendingStream(webrtc::AudioSendStream* stream) {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto count = sending_streams_.erase(stream);
RTC_DCHECK_EQ(1, count);
UpdateAudioTransportWithSendingStreams();
if (sending_streams_.empty()) {
config_.audio_device_module->StopRecording();
}
}
void AudioState::SetPlayout(bool enabled) {
RTC_LOG(LS_INFO) << "SetPlayout(" << enabled << ")";
RTC_DCHECK_RUN_ON(&thread_checker_);
if (playout_enabled_ != enabled) {
playout_enabled_ = enabled;
if (enabled) {
UpdateNullAudioPollerState();
if (!receiving_streams_.empty()) {
config_.audio_device_module->StartPlayout();
}
} else {
config_.audio_device_module->StopPlayout();
UpdateNullAudioPollerState();
}
}
}
void AudioState::SetRecording(bool enabled) {
RTC_LOG(LS_INFO) << "SetRecording(" << enabled << ")";
RTC_DCHECK_RUN_ON(&thread_checker_);
if (recording_enabled_ != enabled) {
recording_enabled_ = enabled;
if (enabled) {
if (!sending_streams_.empty()) {
config_.audio_device_module->StartRecording();
}
} else {
config_.audio_device_module->StopRecording();
}
}
}
void AudioState::SetStereoChannelSwapping(bool enable) {
RTC_DCHECK(thread_checker_.IsCurrent());
audio_transport_.SetStereoChannelSwapping(enable);
}
void AudioState::UpdateAudioTransportWithSendingStreams() {
RTC_DCHECK(thread_checker_.IsCurrent());
std::vector<AudioSender*> audio_senders;
int max_sample_rate_hz = 8000;
size_t max_num_channels = 1;
for (const auto& kv : sending_streams_) {
audio_senders.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_.UpdateAudioSenders(std::move(audio_senders),
max_sample_rate_hz, max_num_channels);
}
void AudioState::UpdateNullAudioPollerState() {
// Run NullAudioPoller when there are receiving streams and playout is
// disabled.
if (!receiving_streams_.empty() && !playout_enabled_) {
if (!null_audio_poller_.Running()) {
AudioTransport* audio_transport = &audio_transport_;
null_audio_poller_ = RepeatingTaskHandle::Start(
TaskQueueBase::Current(), [audio_transport] {
static constexpr size_t kNumChannels = 1;
static constexpr uint32_t kSamplesPerSecond = 48'000;
// 10ms of samples
static constexpr size_t kNumSamples = kSamplesPerSecond / 100;
// Buffer to hold the audio samples.
int16_t buffer[kNumSamples * kNumChannels];
// Output variables from `NeedMorePlayData`.
size_t n_samples;
int64_t elapsed_time_ms;
int64_t ntp_time_ms;
audio_transport->NeedMorePlayData(
kNumSamples, sizeof(int16_t), kNumChannels, kSamplesPerSecond,
buffer, n_samples, &elapsed_time_ms, &ntp_time_ms);
// Reschedule the next poll iteration.
return TimeDelta::Millis(10);
});
}
} else {
null_audio_poller_.Stop();
}
}
} // namespace internal
rtc::scoped_refptr<AudioState> AudioState::Create(
const AudioState::Config& config) {
return rtc::make_ref_counted<internal::AudioState>(config);
}
} // namespace webrtc
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