1 files changed, 151 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
new file mode 100644
index 0000000000..9fbf42ceeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2014 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 "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+
+#include <algorithm>
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kSampleRateHz = 8000;
+
+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
+
+AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
+                                           int payload_type)
+    : frame_size_ms_(config.frame_size_ms),
+      payload_type_(payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      encoder_(nullptr) {
+  RTC_CHECK(config.IsOk());
+  Reset();
+}
+
+AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+}
+
+int AudioEncoderIlbcImpl::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+size_t AudioEncoderIlbcImpl::NumChannels() const {
+  return 1;
+}
+
+size_t AudioEncoderIlbcImpl::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderIlbcImpl::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderIlbcImpl::GetTargetBitrate() const {
+  return GetIlbcBitrate(rtc::dchecked_cast<int>(num_10ms_frames_per_packet_) *
+                        10);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  // Save timestamp if starting a new packet.
+  if (num_10ms_frames_buffered_ == 0)
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  // Buffer input.
+  std::copy(audio.cbegin(), audio.cend(),
+            input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
+
+  // If we don't yet have enough buffered input for a whole packet, we're done
+  // for now.
+  if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+    return EncodedInfo();
+  }
+
+  // Encode buffered input.
+  RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  num_10ms_frames_buffered_ = 0;
+  size_t encoded_bytes = encoded->AppendData(
+      RequiredOutputSizeBytes(), [&](rtc::ArrayView<uint8_t> encoded) {
+        const int r = WebRtcIlbcfix_Encode(
+            encoder_, input_buffer_,
+            kSampleRateHz / 100 * num_10ms_frames_per_packet_, encoded.data());
+        RTC_CHECK_GE(r, 0);
+
+        return static_cast<size_t>(r);
+      });
+
+  RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
+
+  EncodedInfo info;
+  info.encoded_bytes = encoded_bytes;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kIlbc;
+  return info;
+}
+
+void AudioEncoderIlbcImpl::Reset() {
+  if (encoder_)
+    RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
+  const int encoder_frame_size_ms =
+      frame_size_ms_ > 30 ? frame_size_ms_ / 2 : frame_size_ms_;
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+  num_10ms_frames_buffered_ = 0;
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderIlbcImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
+  switch (num_10ms_frames_per_packet_) {
+    case 2:
+      return 38;
+    case 3:
+      return 50;
+    case 4:
+      return 2 * 38;
+    case 6:
+      return 2 * 50;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
+}  // namespace webrtc