Support for Signal calls.

Merge in RedPhone

// FREEBIE

jni/webrtc/modules/audio_coding/main/acm2/acm_opus.cc

@@ -0,0 +1,275 @@
+/*
+ *  Copyright (c) 2012 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/main/acm2/acm_opus.h"
+
+#ifdef WEBRTC_CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#endif
+
+namespace webrtc {
+
+namespace acm2 {
+
+#ifndef WEBRTC_CODEC_OPUS
+
+ACMOpus::ACMOpus(int16_t /* codec_id */)
+    : encoder_inst_ptr_(NULL),
+      sample_freq_(0),
+      bitrate_(0),
+      channels_(1),
+      fec_enabled_(false),
+      packet_loss_rate_(0) {
+  return;
+}
+
+ACMOpus::~ACMOpus() {
+  return;
+}
+
+int16_t ACMOpus::InternalEncode(uint8_t* /* bitstream */,
+                                int16_t* /* bitstream_len_byte */) {
+  return -1;
+}
+
+int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* /* codec_params */) {
+  return -1;
+}
+
+ACMGenericCodec* ACMOpus::CreateInstance(void) {
+  return NULL;
+}
+
+int16_t ACMOpus::InternalCreateEncoder() {
+  return -1;
+}
+
+void ACMOpus::DestructEncoderSafe() {
+  return;
+}
+
+void ACMOpus::InternalDestructEncoderInst(void* /* ptr_inst */) {
+  return;
+}
+
+int16_t ACMOpus::SetBitRateSafe(const int32_t /*rate*/) {
+  return -1;
+}
+
+#else  //===================== Actual Implementation =======================
+
+ACMOpus::ACMOpus(int16_t codec_id)
+    : encoder_inst_ptr_(NULL),
+      sample_freq_(32000),  // Default sampling frequency.
+      bitrate_(20000),  // Default bit-rate.
+      channels_(1),  // Default mono.
+      fec_enabled_(false),  // Default FEC is off.
+      packet_loss_rate_(0) {  // Initial packet loss rate.
+  codec_id_ = codec_id;
+  // Opus has internal DTX, but we dont use it for now.
+  has_internal_dtx_ = false;
+
+  has_internal_fec_ = true;
+
+  if (codec_id_ != ACMCodecDB::kOpus) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
+                 "Wrong codec id for Opus.");
+    sample_freq_ = 0xFFFF;
+    bitrate_ = -1;
+  }
+  return;
+}
+
+ACMOpus::~ACMOpus() {
+  if (encoder_inst_ptr_ != NULL) {
+    WebRtcOpus_EncoderFree(encoder_inst_ptr_);
+    encoder_inst_ptr_ = NULL;
+  }
+}
+
+int16_t ACMOpus::InternalEncode(uint8_t* bitstream,
+                                int16_t* bitstream_len_byte) {
+  // Call Encoder.
+  *bitstream_len_byte = WebRtcOpus_Encode(encoder_inst_ptr_,
+                                          &in_audio_[in_audio_ix_read_],
+                                          frame_len_smpl_,
+                                          MAX_PAYLOAD_SIZE_BYTE, bitstream);
+  // Check for error reported from encoder.
+  if (*bitstream_len_byte < 0) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
+                 "InternalEncode: Encode error for Opus");
+    *bitstream_len_byte = 0;
+    return -1;
+  }
+
+  // Increment the read index. This tells the caller how far
+  // we have gone forward in reading the audio buffer.
+  in_audio_ix_read_ += frame_len_smpl_ * channels_;
+
+  return *bitstream_len_byte;
+}
+
+int16_t ACMOpus::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
+  int16_t ret;
+  if (encoder_inst_ptr_ != NULL) {
+    WebRtcOpus_EncoderFree(encoder_inst_ptr_);
+    encoder_inst_ptr_ = NULL;
+  }
+  ret = WebRtcOpus_EncoderCreate(&encoder_inst_ptr_,
+                                 codec_params->codec_inst.channels);
+  // Store number of channels.
+  channels_ = codec_params->codec_inst.channels;
+
+  if (ret < 0) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
+                 "Encoder creation failed for Opus");
+    return ret;
+  }
+  ret = WebRtcOpus_SetBitRate(encoder_inst_ptr_,
+                              codec_params->codec_inst.rate);
+  if (ret < 0) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
+                 "Setting initial bitrate failed for Opus");
+    return ret;
+  }
+
+  // Store bitrate.
+  bitrate_ = codec_params->codec_inst.rate;
+
+  // TODO(tlegrand): Remove this code when we have proper APIs to set the
+  // complexity at a higher level.
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
+  // If we are on Android, iOS and/or ARM, use a lower complexity setting as
+  // default, to save encoder complexity.
+  const int kOpusComplexity5 = 5;
+  WebRtcOpus_SetComplexity(encoder_inst_ptr_, kOpusComplexity5);
+  if (ret < 0) {
+     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
+                  "Setting complexity failed for Opus");
+     return ret;
+   }
+#endif
+
+  return 0;
+}
+
+ACMGenericCodec* ACMOpus::CreateInstance(void) {
+  return NULL;
+}
+
+int16_t ACMOpus::InternalCreateEncoder() {
+  // Real encoder will be created in InternalInitEncoder.
+  return 0;
+}
+
+void ACMOpus::DestructEncoderSafe() {
+  if (encoder_inst_ptr_) {
+    WebRtcOpus_EncoderFree(encoder_inst_ptr_);
+    encoder_inst_ptr_ = NULL;
+  }
+}
+
+void ACMOpus::InternalDestructEncoderInst(void* ptr_inst) {
+  if (ptr_inst != NULL) {
+    WebRtcOpus_EncoderFree(static_cast<OpusEncInst*>(ptr_inst));
+  }
+  return;
+}
+
+int16_t ACMOpus::SetBitRateSafe(const int32_t rate) {
+  if (rate < 6000 || rate > 510000) {
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
+                 "SetBitRateSafe: Invalid rate Opus");
+    return -1;
+  }
+
+  bitrate_ = rate;
+
+  // Ask the encoder for the new rate.
+  if (WebRtcOpus_SetBitRate(encoder_inst_ptr_, bitrate_) >= 0) {
+    encoder_params_.codec_inst.rate = bitrate_;
+    return 0;
+  }
+
+  return -1;
+}
+
+int ACMOpus::SetFEC(bool enable_fec) {
+  // Ask the encoder to enable FEC.
+  if (enable_fec) {
+    if (WebRtcOpus_EnableFec(encoder_inst_ptr_) == 0) {
+      fec_enabled_ = true;
+      return 0;
+    }
+  } else {
+    if (WebRtcOpus_DisableFec(encoder_inst_ptr_) == 0) {
+      fec_enabled_ = false;
+      return 0;
+    }
+  }
+  return -1;
+}
+
+int ACMOpus::SetPacketLossRate(int loss_rate) {
+  // Optimize the loss rate to configure Opus. Basically, optimized loss rate is
+  // the input loss rate rounded down to various levels, because a robustly good
+  // audio quality is achieved by lowering the packet loss down.
+  // Additionally, to prevent toggling, margins are used, i.e., when jumping to
+  // a loss rate from below, a higher threshold is used than jumping to the same
+  // level from above.
+  const int kPacketLossRate20 = 20;
+  const int kPacketLossRate10 = 10;
+  const int kPacketLossRate5 = 5;
+  const int kPacketLossRate1 = 1;
+  const int kLossRate20Margin = 2;
+  const int kLossRate10Margin = 1;
+  const int kLossRate5Margin = 1;
+  int opt_loss_rate;
+  if (loss_rate >= kPacketLossRate20 + kLossRate20Margin *
+      (kPacketLossRate20 - packet_loss_rate_ > 0 ? 1 : -1)) {
+    opt_loss_rate = kPacketLossRate20;
+  } else if (loss_rate >= kPacketLossRate10 + kLossRate10Margin *
+      (kPacketLossRate10 - packet_loss_rate_ > 0 ? 1 : -1)) {
+    opt_loss_rate = kPacketLossRate10;
+  } else if (loss_rate >= kPacketLossRate5 + kLossRate5Margin *
+      (kPacketLossRate5 - packet_loss_rate_ > 0 ? 1 : -1)) {
+    opt_loss_rate = kPacketLossRate5;
+  } else if (loss_rate >= kPacketLossRate1) {
+    opt_loss_rate = kPacketLossRate1;
+  } else {
+    opt_loss_rate = 0;
+  }
+
+  if (packet_loss_rate_ == opt_loss_rate) {
+    return 0;
+  }
+
+  // Ask the encoder to change the target packet loss rate.
+  if (WebRtcOpus_SetPacketLossRate(encoder_inst_ptr_, opt_loss_rate) == 0) {
+    packet_loss_rate_ = opt_loss_rate;
+    return 0;
+  }
+
+  return -1;
+}
+
+int ACMOpus::SetOpusMaxBandwidth(int max_bandwidth) {
+  // Ask the encoder to change the maximum required bandwidth.
+  return WebRtcOpus_SetMaxBandwidth(encoder_inst_ptr_, max_bandwidth);
+}
+
+#endif  // WEBRTC_CODEC_OPUS
+
+}  // namespace acm2
+
+}  // namespace webrtc