Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 212 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/audio_state.cc b/third_party/libwebrtc/audio/audio_state.cc
new file mode 100644
index 0000000000..6f20e7b128
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_state.cc
@@ -0,0 +1,212 @@
+/*
+ *  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