1 files changed, 86 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc
new file mode 100644
index 0000000000..efc9a73546
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc
@@ -0,0 +1,86 @@
+/*
+ *  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.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+
+namespace webrtc {
+
+bool AudioDecoderOpus::Config::IsOk() const {
+  if (sample_rate_hz != 16000 && sample_rate_hz != 48000) {
+    // Unsupported sample rate. (libopus supports a few other rates as
+    // well; we can add support for them when needed.)
+    return false;
+  }
+  if (num_channels != 1 && num_channels != 2) {
+    return false;
+  }
+  return true;
+}
+
+absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const auto num_channels = [&]() -> absl::optional<int> {
+    auto stereo = format.parameters.find("stereo");
+    if (stereo != format.parameters.end()) {
+      if (stereo->second == "0") {
+        return 1;
+      } else if (stereo->second == "1") {
+        return 2;
+      } else {
+        return absl::nullopt;  // Bad stereo parameter.
+      }
+    }
+    return 1;  // Default to mono.
+  }();
+  if (absl::EqualsIgnoreCase(format.name, "opus") &&
+      format.clockrate_hz == 48000 && format.num_channels == 2 &&
+      num_channels) {
+    Config config;
+    config.num_channels = *num_channels;
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioDecoderOpus::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
+  opus_info.allow_comfort_noise = false;
+  opus_info.supports_network_adaption = true;
+  SdpAudioFormat opus_format(
+      {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}});
+  specs->push_back({std::move(opus_format), opus_info});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioDecoderOpusImpl>(config.num_channels,
+                                                config.sample_rate_hz);
+}
+
+}  // namespace webrtc