1 files changed, 177 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device_defines.h b/third_party/libwebrtc/modules/audio_device/include/audio_device_defines.h
new file mode 100644
index 0000000000..d677d41f69
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+++ b/third_party/libwebrtc/modules/audio_device/include/audio_device_defines.h
@@ -0,0 +1,177 @@
+/*
+ *  Copyright (c) 2011 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 MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+
+#include <stddef.h>
+
+#include <string>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+static const int kAdmMaxDeviceNameSize = 128;
+static const int kAdmMaxFileNameSize = 512;
+static const int kAdmMaxGuidSize = 128;
+
+static const int kAdmMinPlayoutBufferSizeMs = 10;
+static const int kAdmMaxPlayoutBufferSizeMs = 250;
+
+// ----------------------------------------------------------------------------
+//  AudioTransport
+// ----------------------------------------------------------------------------
+
+class AudioTransport {
+ public:
+  // TODO(bugs.webrtc.org/13620) Deprecate this function
+  virtual 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) = 0;  // NOLINT
+
+  virtual 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> estimatedCaptureTimeNS) {  // NOLINT
+    // TODO(webrtc:13620) Make the default behaver of the new API to behave as
+    // the old API. This can be pure virtual if all uses of the old API is
+    // removed.
+    return RecordedDataIsAvailable(
+        audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
+        totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
+  }
+
+  // Implementation has to setup safe values for all specified out parameters.
+  virtual int32_t NeedMorePlayData(size_t nSamples,
+                                   size_t nBytesPerSample,
+                                   size_t nChannels,
+                                   uint32_t samplesPerSec,
+                                   void* audioSamples,
+                                   size_t& nSamplesOut,  // NOLINT
+                                   int64_t* elapsed_time_ms,
+                                   int64_t* ntp_time_ms) = 0;  // NOLINT
+
+  // Method to pull mixed render audio data from all active VoE channels.
+  // The data will not be passed as reference for audio processing internally.
+  virtual 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) = 0;
+
+ protected:
+  virtual ~AudioTransport() {}
+};
+
+// Helper class for storage of fundamental audio parameters such as sample rate,
+// number of channels, native buffer size etc.
+// Note that one audio frame can contain more than one channel sample and each
+// sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
+// stereo contains 2 * (16/8) = 4 bytes of data.
+class AudioParameters {
+ public:
+  // This implementation does only support 16-bit PCM samples.
+  static const size_t kBitsPerSample = 16;
+  AudioParameters()
+      : sample_rate_(0),
+        channels_(0),
+        frames_per_buffer_(0),
+        frames_per_10ms_buffer_(0) {}
+  AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
+      : sample_rate_(sample_rate),
+        channels_(channels),
+        frames_per_buffer_(frames_per_buffer),
+        frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
+  void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
+    sample_rate_ = sample_rate;
+    channels_ = channels;
+    frames_per_buffer_ = frames_per_buffer;
+    frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
+  }
+  size_t bits_per_sample() const { return kBitsPerSample; }
+  void reset(int sample_rate, size_t channels, double buffer_duration) {
+    reset(sample_rate, channels,
+          static_cast<size_t>(sample_rate * buffer_duration + 0.5));
+  }
+  void reset(int sample_rate, size_t channels) {
+    reset(sample_rate, channels, static_cast<size_t>(0));
+  }
+  int sample_rate() const { return sample_rate_; }
+  size_t channels() const { return channels_; }
+  size_t frames_per_buffer() const { return frames_per_buffer_; }
+  size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
+  size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
+  size_t GetBytesPerBuffer() const {
+    return frames_per_buffer_ * GetBytesPerFrame();
+  }
+  // The WebRTC audio device buffer (ADB) only requires that the sample rate
+  // and number of channels are configured. Hence, to be "valid", only these
+  // two attributes must be set.
+  bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); }
+  // Most platforms also require that a native buffer size is defined.
+  // An audio parameter instance is considered to be "complete" if it is both
+  // "valid" (can be used by the ADB) and also has a native frame size.
+  bool is_complete() const { return (is_valid() && (frames_per_buffer_ > 0)); }
+  size_t GetBytesPer10msBuffer() const {
+    return frames_per_10ms_buffer_ * GetBytesPerFrame();
+  }
+  double GetBufferSizeInMilliseconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return frames_per_buffer_ / (sample_rate_ / 1000.0);
+  }
+  double GetBufferSizeInSeconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return static_cast<double>(frames_per_buffer_) / (sample_rate_);
+  }
+  std::string ToString() const {
+    char ss_buf[1024];
+    rtc::SimpleStringBuilder ss(ss_buf);
+    ss << "AudioParameters: ";
+    ss << "sample_rate=" << sample_rate() << ", channels=" << channels();
+    ss << ", frames_per_buffer=" << frames_per_buffer();
+    ss << ", frames_per_10ms_buffer=" << frames_per_10ms_buffer();
+    ss << ", bytes_per_frame=" << GetBytesPerFrame();
+    ss << ", bytes_per_buffer=" << GetBytesPerBuffer();
+    ss << ", bytes_per_10ms_buffer=" << GetBytesPer10msBuffer();
+    ss << ", size_in_ms=" << GetBufferSizeInMilliseconds();
+    return ss.str();
+  }
+
+ private:
+  int sample_rate_;
+  size_t channels_;
+  size_t frames_per_buffer_;
+  size_t frames_per_10ms_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_