1 files changed, 164 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
new file mode 100644
index 0000000000..43b097fa0e
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -0,0 +1,164 @@
+/*
+ *  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 "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h"
+
+#include <limits>
+#include "webrtc/base/checks.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kSampleRateHz = 16000;
+
+AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
+  AudioEncoderG722::Config config;
+  config.num_channels = codec_inst.channels;
+  config.frame_size_ms = codec_inst.pacsize / 16;
+  config.payload_type = codec_inst.pltype;
+  return config;
+}
+
+}  // namespace
+
+bool AudioEncoderG722::Config::IsOk() const {
+  return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) &&
+      (num_channels >= 1);
+}
+
+AudioEncoderG722::AudioEncoderG722(const Config& config)
+    : num_channels_(config.num_channels),
+      payload_type_(config.payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      num_10ms_frames_buffered_(0),
+      first_timestamp_in_buffer_(0),
+      encoders_(new EncoderState[num_channels_]),
+      interleave_buffer_(2 * num_channels_) {
+  RTC_CHECK(config.IsOk());
+  const size_t samples_per_channel =
+      kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+  for (int i = 0; i < num_channels_; ++i) {
+    encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
+    encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
+  }
+  Reset();
+}
+
+AudioEncoderG722::AudioEncoderG722(const CodecInst& codec_inst)
+    : AudioEncoderG722(CreateConfig(codec_inst)) {}
+
+AudioEncoderG722::~AudioEncoderG722() = default;
+
+size_t AudioEncoderG722::MaxEncodedBytes() const {
+  return SamplesPerChannel() / 2 * num_channels_;
+}
+
+int AudioEncoderG722::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+int AudioEncoderG722::NumChannels() const {
+  return num_channels_;
+}
+
+int AudioEncoderG722::RtpTimestampRateHz() const {
+  // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
+  // codec.
+  return kSampleRateHz / 2;
+}
+
+size_t AudioEncoderG722::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderG722::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderG722::GetTargetBitrate() const {
+  // 4 bits/sample, 16000 samples/s/channel.
+  return 64000 * NumChannels();
+}
+
+AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
+    uint32_t rtp_timestamp,
+    const int16_t* audio,
+    size_t max_encoded_bytes,
+    uint8_t* encoded) {
+  RTC_CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
+
+  if (num_10ms_frames_buffered_ == 0)
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  // Deinterleave samples and save them in each channel's buffer.
+  const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
+  for (size_t i = 0; i < kSampleRateHz / 100; ++i)
+    for (int j = 0; j < num_channels_; ++j)
+      encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
+
+  // If we don't yet have enough samples for a packet, we're done for now.
+  if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+    return EncodedInfo();
+  }
+
+  // Encode each channel separately.
+  RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  num_10ms_frames_buffered_ = 0;
+  const size_t samples_per_channel = SamplesPerChannel();
+  for (int i = 0; i < num_channels_; ++i) {
+    const size_t encoded = WebRtcG722_Encode(
+        encoders_[i].encoder, encoders_[i].speech_buffer.get(),
+        samples_per_channel, encoders_[i].encoded_buffer.data());
+    RTC_CHECK_EQ(encoded, samples_per_channel / 2);
+  }
+
+  // Interleave the encoded bytes of the different channels. Each separate
+  // channel and the interleaved stream encodes two samples per byte, most
+  // significant half first.
+  for (size_t i = 0; i < samples_per_channel / 2; ++i) {
+    for (int j = 0; j < num_channels_; ++j) {
+      uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
+      interleave_buffer_.data()[j] = two_samples >> 4;
+      interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
+    }
+    for (int j = 0; j < num_channels_; ++j)
+      encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 |
+                                       interleave_buffer_.data()[2 * j + 1];
+  }
+  EncodedInfo info;
+  info.encoded_bytes = samples_per_channel / 2 * num_channels_;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  return info;
+}
+
+void AudioEncoderG722::Reset() {
+  num_10ms_frames_buffered_ = 0;
+  for (int i = 0; i < num_channels_; ++i)
+    RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
+}
+
+AudioEncoderG722::EncoderState::EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
+}
+
+AudioEncoderG722::EncoderState::~EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
+}
+
+size_t AudioEncoderG722::SamplesPerChannel() const {
+  return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+}
+
+}  // namespace webrtc