Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 100 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestStereo.h b/third_party/libwebrtc/modules/audio_coding/test/TestStereo.h
new file mode 100644
index 0000000000..4c50a4b555
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestStereo.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2012 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_CODING_TEST_TESTSTEREO_H_
+#define MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+
+#include <math.h>
+
+#include <memory>
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/PCMFile.h"
+
+#define PCMA_AND_PCMU
+
+namespace webrtc {
+
+enum StereoMonoMode { kNotSet, kMono, kStereo };
+
+class TestPackStereo : public AudioPacketizationCallback {
+ public:
+  TestPackStereo();
+  ~TestPackStereo();
+
+  void RegisterReceiverACM(AudioCodingModule* acm);
+
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_size,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  uint16_t payload_size();
+  uint32_t timestamp_diff();
+  void reset_payload_size();
+  void set_codec_mode(StereoMonoMode mode);
+  void set_lost_packet(bool lost);
+
+ private:
+  AudioCodingModule* receiver_acm_;
+  int16_t seq_no_;
+  uint32_t timestamp_diff_;
+  uint32_t last_in_timestamp_;
+  uint64_t total_bytes_;
+  int payload_size_;
+  StereoMonoMode codec_mode_;
+  // Simulate packet losses
+  bool lost_packet_;
+};
+
+class TestStereo {
+ public:
+  TestStereo();
+  ~TestStereo();
+
+  void Perform();
+
+ private:
+  // The default value of '-1' indicates that the registration is based only on
+  // codec name and a sampling frequncy matching is not required. This is useful
+  // for codecs which support several sampling frequency.
+  void RegisterSendCodec(char side,
+                         char* codec_name,
+                         int32_t samp_freq_hz,
+                         int rate,
+                         int pack_size,
+                         int channels);
+
+  void Run(TestPackStereo* channel,
+           int in_channels,
+           int out_channels,
+           int percent_loss = 0);
+  void OpenOutFile(int16_t test_number);
+
+  std::unique_ptr<AudioCodingModule> acm_a_;
+  std::unique_ptr<AudioCodingModule> acm_b_;
+
+  TestPackStereo* channel_a2b_;
+
+  PCMFile* in_file_stereo_;
+  PCMFile* in_file_mono_;
+  PCMFile out_file_;
+  int16_t test_cntr_;
+  uint16_t pack_size_samp_;
+  uint16_t pack_size_bytes_;
+  int counter_;
+  char* send_codec_name_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_