1 files changed, 190 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
new file mode 100644
index 0000000..279f80d
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -0,0 +1,190 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
+
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+template <typename T>
+typename AudioEncoderIsacT<T>::Config CreateIsacConfig(
+    const CodecInst& codec_inst,
+    LockedIsacBandwidthInfo* bwinfo) {
+  typename AudioEncoderIsacT<T>::Config config;
+  config.bwinfo = bwinfo;
+  config.payload_type = codec_inst.pltype;
+  config.sample_rate_hz = codec_inst.plfreq;
+  config.frame_size_ms =
+      rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz);
+  config.adaptive_mode = (codec_inst.rate == -1);
+  if (codec_inst.rate != -1)
+    config.bit_rate = codec_inst.rate;
+  return config;
+}
+
+template <typename T>
+bool AudioEncoderIsacT<T>::Config::IsOk() const {
+  if (max_bit_rate < 32000 && max_bit_rate != -1)
+    return false;
+  if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
+    return false;
+  if (adaptive_mode && !bwinfo)
+    return false;
+  switch (sample_rate_hz) {
+    case 16000:
+      if (max_bit_rate > 53400)
+        return false;
+      if (max_payload_size_bytes > 400)
+        return false;
+      return (frame_size_ms == 30 || frame_size_ms == 60) &&
+             (bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 32000));
+    case 32000:
+      if (max_bit_rate > 160000)
+        return false;
+      if (max_payload_size_bytes > 600)
+        return false;
+      return T::has_swb &&
+             (frame_size_ms == 30 &&
+              (bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 56000)));
+    default:
+      return false;
+  }
+}
+
+template <typename T>
+AudioEncoderIsacT<T>::AudioEncoderIsacT(const Config& config) {
+  RecreateEncoderInstance(config);
+}
+
+template <typename T>
+AudioEncoderIsacT<T>::AudioEncoderIsacT(const CodecInst& codec_inst,
+                                        LockedIsacBandwidthInfo* bwinfo)
+    : AudioEncoderIsacT(CreateIsacConfig<T>(codec_inst, bwinfo)) {}
+
+template <typename T>
+AudioEncoderIsacT<T>::~AudioEncoderIsacT() {
+  RTC_CHECK_EQ(0, T::Free(isac_state_));
+}
+
+template <typename T>
+size_t AudioEncoderIsacT<T>::MaxEncodedBytes() const {
+  return kSufficientEncodeBufferSizeBytes;
+}
+
+template <typename T>
+int AudioEncoderIsacT<T>::SampleRateHz() const {
+  return T::EncSampRate(isac_state_);
+}
+
+template <typename T>
+int AudioEncoderIsacT<T>::NumChannels() const {
+  return 1;
+}
+
+template <typename T>
+size_t AudioEncoderIsacT<T>::Num10MsFramesInNextPacket() const {
+  const int samples_in_next_packet = T::GetNewFrameLen(isac_state_);
+  return static_cast<size_t>(
+      rtc::CheckedDivExact(samples_in_next_packet,
+                           rtc::CheckedDivExact(SampleRateHz(), 100)));
+}
+
+template <typename T>
+size_t AudioEncoderIsacT<T>::Max10MsFramesInAPacket() const {
+  return 6;  // iSAC puts at most 60 ms in a packet.
+}
+
+template <typename T>
+int AudioEncoderIsacT<T>::GetTargetBitrate() const {
+  if (config_.adaptive_mode)
+    return -1;
+  return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate;
+}
+
+template <typename T>
+AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeInternal(
+    uint32_t rtp_timestamp,
+    const int16_t* audio,
+    size_t max_encoded_bytes,
+    uint8_t* encoded) {
+  if (!packet_in_progress_) {
+    // Starting a new packet; remember the timestamp for later.
+    packet_in_progress_ = true;
+    packet_timestamp_ = rtp_timestamp;
+  }
+  if (bwinfo_) {
+    IsacBandwidthInfo bwinfo = bwinfo_->Get();
+    T::SetBandwidthInfo(isac_state_, &bwinfo);
+  }
+  int r = T::Encode(isac_state_, audio, encoded);
+  RTC_CHECK_GE(r, 0) << "Encode failed (error code "
+                     << T::GetErrorCode(isac_state_) << ")";
+
+  // T::Encode doesn't allow us to tell it the size of the output
+  // buffer. All we can do is check for an overrun after the fact.
+  RTC_CHECK_LE(static_cast<size_t>(r), max_encoded_bytes);
+
+  if (r == 0)
+    return EncodedInfo();
+
+  // Got enough input to produce a packet. Return the saved timestamp from
+  // the first chunk of input that went into the packet.
+  packet_in_progress_ = false;
+  EncodedInfo info;
+  info.encoded_bytes = r;
+  info.encoded_timestamp = packet_timestamp_;
+  info.payload_type = config_.payload_type;
+  return info;
+}
+
+template <typename T>
+void AudioEncoderIsacT<T>::Reset() {
+  RecreateEncoderInstance(config_);
+}
+
+template <typename T>
+void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
+  RTC_CHECK(config.IsOk());
+  packet_in_progress_ = false;
+  bwinfo_ = config.bwinfo;
+  if (isac_state_)
+    RTC_CHECK_EQ(0, T::Free(isac_state_));
+  RTC_CHECK_EQ(0, T::Create(&isac_state_));
+  RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
+  RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
+  const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
+  if (config.adaptive_mode) {
+    RTC_CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate, config.frame_size_ms,
+                                  config.enforce_frame_size));
+  } else {
+    RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
+  }
+  if (config.max_payload_size_bytes != -1)
+    RTC_CHECK_EQ(
+        0, T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
+  if (config.max_bit_rate != -1)
+    RTC_CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
+
+  // Set the decoder sample rate even though we just use the encoder. This
+  // doesn't appear to be necessary to produce a valid encoding, but without it
+  // we get an encoding that isn't bit-for-bit identical with what a combined
+  // encoder+decoder object produces.
+  RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
+
+  config_ = config;
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_