Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

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 0000000000..279f80d6fc
--- /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_