1 files changed, 54 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h
new file mode 100644
index 0000000000..47509849de
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+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2017 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 API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
+#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// L16 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderL16 {
+  struct Config {
+    bool IsOk() const {
+      return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+              sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
+             num_channels >= 1 &&
+             num_channels <= AudioEncoder::kMaxNumberOfChannels &&
+             frame_size_ms > 0 && frame_size_ms <= 120 &&
+             frame_size_ms % 10 == 0;
+    }
+    int sample_rate_hz = 8000;
+    int num_channels = 1;
+    int frame_size_ms = 10;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_