Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 64 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
new file mode 100644
index 0000000000..e8fd0440bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -0,0 +1,64 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoderOpusImpl final : public AudioDecoder {
+ public:
+  explicit AudioDecoderOpusImpl(size_t num_channels,
+                                int sample_rate_hz = 48000);
+  ~AudioDecoderOpusImpl() override;
+
+  AudioDecoderOpusImpl(const AudioDecoderOpusImpl&) = delete;
+  AudioDecoderOpusImpl& operator=(const AudioDecoderOpusImpl&) = delete;
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  OpusDecInst* dec_state_;
+  const size_t channels_;
+  const int sample_rate_hz_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_