1 files changed, 182 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
new file mode 100644
index 0000000000..285ea89959
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2019 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/opus/audio_decoder_multi_channel_opus_impl.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+std::unique_ptr<AudioDecoderMultiChannelOpusImpl>
+AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(
+    AudioDecoderMultiChannelOpusConfig config) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  // Fill the pointer with a working decoder through the C interface. This
+  // allocates memory.
+  OpusDecInst* dec_state = nullptr;
+  const int error = WebRtcOpus_MultistreamDecoderCreate(
+      &dec_state, config.num_channels, config.num_streams,
+      config.coupled_streams, config.channel_mapping.data());
+  if (error != 0) {
+    return nullptr;
+  }
+
+  // Pass the ownership to DecoderImpl. Not using 'make_unique' because the
+  // c-tor is private.
+  return std::unique_ptr<AudioDecoderMultiChannelOpusImpl>(
+      new AudioDecoderMultiChannelOpusImpl(dec_state, config));
+}
+
+AudioDecoderMultiChannelOpusImpl::AudioDecoderMultiChannelOpusImpl(
+    OpusDecInst* dec_state,
+    AudioDecoderMultiChannelOpusConfig config)
+    : dec_state_(dec_state), config_(config) {
+  RTC_DCHECK(dec_state);
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderMultiChannelOpusImpl::~AudioDecoderMultiChannelOpusImpl() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+absl::optional<AudioDecoderMultiChannelOpusConfig>
+AudioDecoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) {
+  AudioDecoderMultiChannelOpusConfig config;
+  config.num_channels = format.num_channels;
+  auto num_streams = GetFormatParameter<int>(format, "num_streams");
+  if (!num_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.num_streams = *num_streams;
+
+  auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams");
+  if (!coupled_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.coupled_streams = *coupled_streams;
+
+  auto channel_mapping =
+      GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping");
+  if (!channel_mapping.has_value()) {
+    return absl::nullopt;
+  }
+  config.channel_mapping = *channel_mapping;
+  if (!config.IsOk()) {
+    return absl::nullopt;
+  }
+  return config;
+}
+
+std::vector<AudioDecoder::ParseResult>
+AudioDecoderMultiChannelOpusImpl::ParsePayload(rtc::Buffer&& payload,
+                                               uint32_t timestamp) {
+  std::vector<ParseResult> results;
+
+  if (PacketHasFec(payload.data(), payload.size())) {
+    const int duration =
+        PacketDurationRedundant(payload.data(), payload.size());
+    RTC_DCHECK_GE(duration, 0);
+    rtc::Buffer payload_copy(payload.data(), payload.size());
+    std::unique_ptr<EncodedAudioFrame> fec_frame(
+        new OpusFrame(this, std::move(payload_copy), false));
+    results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
+  }
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OpusFrame(this, std::move(payload), true));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoderMultiChannelOpusImpl::DecodeInternal(const uint8_t* encoded,
+                                                     size_t encoded_len,
+                                                     int sample_rate_hz,
+                                                     int16_t* decoded,
+                                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(
+        config_.num_channels);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderMultiChannelOpusImpl::DecodeRedundantInternal(
+    const uint8_t* encoded,
+    size_t encoded_len,
+    int sample_rate_hz,
+    int16_t* decoded,
+    SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(
+        config_.num_channels);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderMultiChannelOpusImpl::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderMultiChannelOpusImpl::PacketDuration(const uint8_t* encoded,
+                                                     size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderMultiChannelOpusImpl::PacketDurationRedundant(
+    const uint8_t* encoded,
+    size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len, 48000);
+}
+
+bool AudioDecoderMultiChannelOpusImpl::PacketHasFec(const uint8_t* encoded,
+                                                    size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+int AudioDecoderMultiChannelOpusImpl::SampleRateHz() const {
+  return 48000;
+}
+
+size_t AudioDecoderMultiChannelOpusImpl::Channels() const {
+  return config_.num_channels;
+}
+
+}  // namespace webrtc