1 files changed, 76 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc
new file mode 100644
index 0000000000..20259b9ad8
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc
@@ -0,0 +1,76 @@
+/*
+ *  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/L16/audio_encoder_L16.h"
+
+#include <memory>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  Config config;
+  config.sample_rate_hz = format.clockrate_hz;
+  config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
+    }
+  }
+  if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
+    return config;
+  }
+  return absl::nullopt;
+}
+
+void AudioEncoderL16::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  Pcm16BAppendSupportedCodecSpecs(specs);
+}
+
+AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
+    const AudioEncoderL16::Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {config.sample_rate_hz,
+          rtc::dchecked_cast<size_t>(config.num_channels),
+          config.sample_rate_hz * config.num_channels * 16};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
+    const AudioEncoderL16::Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  AudioEncoderPcm16B::Config c;
+  c.sample_rate_hz = config.sample_rate_hz;
+  c.num_channels = config.num_channels;
+  c.frame_size_ms = config.frame_size_ms;
+  c.payload_type = payload_type;
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderPcm16B>(c);
+}
+
+}  // namespace webrtc