Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

1 files changed, 117 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/audio_transport_impl.h b/third_party/libwebrtc/audio/audio_transport_impl.h
new file mode 100644
index 0000000000..24b09d2140
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_transport_impl.h
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#ifndef AUDIO_AUDIO_TRANSPORT_IMPL_H_
+#define AUDIO_AUDIO_TRANSPORT_IMPL_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/audio_mixer.h"
+#include "api/scoped_refptr.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "modules/async_audio_processing/async_audio_processing.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioSender;
+
+class AudioTransportImpl : public AudioTransport {
+ public:
+  AudioTransportImpl(
+      AudioMixer* mixer,
+      AudioProcessing* audio_processing,
+      AsyncAudioProcessing::Factory* async_audio_processing_factory);
+
+  AudioTransportImpl() = delete;
+  AudioTransportImpl(const AudioTransportImpl&) = delete;
+  AudioTransportImpl& operator=(const AudioTransportImpl&) = delete;
+
+  ~AudioTransportImpl() override;
+
+  // TODO(bugs.webrtc.org/13620) Deprecate this function
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  size_t nSamples,
+                                  size_t nBytesPerSample,
+                                  size_t nChannels,
+                                  uint32_t samplesPerSec,
+                                  uint32_t totalDelayMS,
+                                  int32_t clockDrift,
+                                  uint32_t currentMicLevel,
+                                  bool keyPressed,
+                                  uint32_t& newMicLevel) override;
+
+  int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t nSamples,
+      size_t nBytesPerSample,
+      size_t nChannels,
+      uint32_t samplesPerSec,
+      uint32_t totalDelayMS,
+      int32_t clockDrift,
+      uint32_t currentMicLevel,
+      bool keyPressed,
+      uint32_t& newMicLevel,
+      absl::optional<int64_t> estimated_capture_time_ns) override;
+
+  int32_t NeedMorePlayData(size_t nSamples,
+                           size_t nBytesPerSample,
+                           size_t nChannels,
+                           uint32_t samplesPerSec,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override;
+
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override;
+
+  void UpdateAudioSenders(std::vector<AudioSender*> senders,
+                          int send_sample_rate_hz,
+                          size_t send_num_channels);
+  void SetStereoChannelSwapping(bool enable);
+
+ private:
+  void SendProcessedData(std::unique_ptr<AudioFrame> audio_frame);
+
+  // Shared.
+  AudioProcessing* audio_processing_ = nullptr;
+
+  // Capture side.
+
+  // Thread-safe.
+  const std::unique_ptr<AsyncAudioProcessing> async_audio_processing_;
+
+  mutable Mutex capture_lock_;
+  std::vector<AudioSender*> audio_senders_ RTC_GUARDED_BY(capture_lock_);
+  int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000;
+  size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
+  bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
+  PushResampler<int16_t> capture_resampler_;
+
+  // Render side.
+
+  rtc::scoped_refptr<AudioMixer> mixer_;
+  AudioFrame mixed_frame_;
+  // Converts mixed audio to the audio device output rate.
+  PushResampler<int16_t> render_resampler_;
+};
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_TRANSPORT_IMPL_H_