1 files changed, 89 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
new file mode 100644
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+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2019 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_CODER_OPUS_COMMON_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+                                               absl::string_view param);
+
+template <typename T>
+absl::optional<T> GetFormatParameter(const SdpAudioFormat& format,
+                                     absl::string_view param) {
+  return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
+}
+
+template <>
+absl::optional<std::vector<unsigned char>> GetFormatParameter(
+    const SdpAudioFormat& format,
+    absl::string_view param);
+
+class OpusFrame : public AudioDecoder::EncodedAudioFrame {
+ public:
+  OpusFrame(AudioDecoder* decoder,
+            rtc::Buffer&& payload,
+            bool is_primary_payload)
+      : decoder_(decoder),
+        payload_(std::move(payload)),
+        is_primary_payload_(is_primary_payload) {}
+
+  size_t Duration() const override {
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    } else {
+      ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
+    }
+    return (ret < 0) ? 0 : static_cast<size_t>(ret);
+  }
+
+  bool IsDtxPacket() const override { return payload_.size() <= 2; }
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->Decode(
+          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    } else {
+      ret = decoder_->DecodeRedundant(
+          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    }
+
+    if (ret < 0)
+      return absl::nullopt;
+
+    return DecodeResult{static_cast<size_t>(ret), speech_type};
+  }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+  const bool is_primary_payload_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_