1 files changed, 88 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
new file mode 100644
index 0000000000..b497948491
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
@@ -0,0 +1,88 @@
+/*
+ *  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/ilbc/audio_encoder_ilbc.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+namespace {
+int GetIlbcBitrate(int ptime) {
+  switch (ptime) {
+    case 20:
+    case 40:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 30:
+    case 60:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+}  // namespace
+
+absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") ||
+      format.clockrate_hz != 8000 || format.num_channels != 1) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderIlbcConfig config;
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      const int whole_packets = *ptime / 10;
+      config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
+    }
+  }
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+void AudioEncoderIlbc::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"ILBC", 8000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder(
+    const AudioEncoderIlbcConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return {8000, 1, GetIlbcBitrate(config.frame_size_ms)};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
+    const AudioEncoderIlbcConfig& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderIlbcImpl>(config, payload_type);
+}
+
+}  // namespace webrtc