Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

2 files changed, 158 insertions, 0 deletions

diff --git a/third_party/libwebrtc/common_audio/resampler/include/push_resampler.h b/third_party/libwebrtc/common_audio/resampler/include/push_resampler.h
new file mode 100644
index 0000000000..3da67120f0
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/resampler/include/push_resampler.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2013 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 COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
+#define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
+
+#include <memory>
+#include <vector>
+
+namespace webrtc {
+
+class PushSincResampler;
+
+// Wraps PushSincResampler to provide stereo support.
+// TODO(ajm): add support for an arbitrary number of channels.
+template <typename T>
+class PushResampler {
+ public:
+  PushResampler();
+  virtual ~PushResampler();
+
+  // Must be called whenever the parameters change. Free to be called at any
+  // time as it is a no-op if parameters have not changed since the last call.
+  int InitializeIfNeeded(int src_sample_rate_hz,
+                         int dst_sample_rate_hz,
+                         size_t num_channels);
+
+  // Returns the total number of samples provided in destination (e.g. 32 kHz,
+  // 2 channel audio gives 640 samples).
+  int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
+
+ private:
+  int src_sample_rate_hz_;
+  int dst_sample_rate_hz_;
+  size_t num_channels_;
+  // Vector that is needed to provide the proper inputs and outputs to the
+  // interleave/de-interleave methods used in Resample. This needs to be
+  // heap-allocated on the state to support an arbitrary number of channels
+  // without doing run-time heap-allocations in the Resample method.
+  std::vector<T*> channel_data_array_;
+
+  struct ChannelResampler {
+    std::unique_ptr<PushSincResampler> resampler;
+    std::vector<T> source;
+    std::vector<T> destination;
+  };
+
+  std::vector<ChannelResampler> channel_resamplers_;
+};
+}  // namespace webrtc
+
+#endif  // COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
diff --git a/third_party/libwebrtc/common_audio/resampler/include/resampler.h b/third_party/libwebrtc/common_audio/resampler/include/resampler.h
new file mode 100644
index 0000000000..41940f9a12
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/resampler/include/resampler.h
@@ -0,0 +1,99 @@
+/*
+ *  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.
+ */
+
+/*
+ * A wrapper for resampling a numerous amount of sampling combinations.
+ */
+
+#ifndef COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
+#define COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// All methods return 0 on success and -1 on failure.
+class Resampler {
+ public:
+  Resampler();
+  Resampler(int inFreq, int outFreq, size_t num_channels);
+  ~Resampler();
+
+  // Reset all states
+  int Reset(int inFreq, int outFreq, size_t num_channels);
+
+  // Reset all states if any parameter has changed
+  int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels);
+
+  // Resample samplesIn to samplesOut.
+  int Push(const int16_t* samplesIn,
+           size_t lengthIn,
+           int16_t* samplesOut,
+           size_t maxLen,
+           size_t& outLen);  // NOLINT: to avoid changing APIs
+
+ private:
+  enum ResamplerMode {
+    kResamplerMode1To1,
+    kResamplerMode1To2,
+    kResamplerMode1To3,
+    kResamplerMode1To4,
+    kResamplerMode1To6,
+    kResamplerMode1To12,
+    kResamplerMode2To3,
+    kResamplerMode2To11,
+    kResamplerMode4To11,
+    kResamplerMode8To11,
+    kResamplerMode11To16,
+    kResamplerMode11To32,
+    kResamplerMode2To1,
+    kResamplerMode3To1,
+    kResamplerMode4To1,
+    kResamplerMode6To1,
+    kResamplerMode12To1,
+    kResamplerMode3To2,
+    kResamplerMode11To2,
+    kResamplerMode11To4,
+    kResamplerMode11To8
+  };
+
+  // Computes the resampler mode for a given sampling frequency pair.
+  // Returns -1 for unsupported frequency pairs.
+  static int ComputeResamplerMode(int in_freq_hz,
+                                  int out_freq_hz,
+                                  ResamplerMode* mode);
+
+  // Generic pointers since we don't know what states we'll need
+  void* state1_;
+  void* state2_;
+  void* state3_;
+
+  // Storage if needed
+  int16_t* in_buffer_;
+  int16_t* out_buffer_;
+  size_t in_buffer_size_;
+  size_t out_buffer_size_;
+  size_t in_buffer_size_max_;
+  size_t out_buffer_size_max_;
+
+  int my_in_frequency_khz_;
+  int my_out_frequency_khz_;
+  ResamplerMode my_mode_;
+  size_t num_channels_;
+
+  // Extra instance for stereo
+  Resampler* helper_left_;
+  Resampler* helper_right_;
+};
+
+}  // namespace webrtc
+
+#endif  // COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_