Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 92 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
new file mode 100644
index 0000000000..8a7210515c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
@@ -0,0 +1,92 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class AudioEncoderMultiChannelOpusImpl final : public AudioEncoder {
+ public:
+  AudioEncoderMultiChannelOpusImpl(
+      const AudioEncoderMultiChannelOpusConfig& config,
+      int payload_type);
+  ~AudioEncoderMultiChannelOpusImpl() override;
+
+  AudioEncoderMultiChannelOpusImpl(const AudioEncoderMultiChannelOpusImpl&) =
+      delete;
+  AudioEncoderMultiChannelOpusImpl& operator=(
+      const AudioEncoderMultiChannelOpusImpl&) = delete;
+
+  // Static interface for use by BuiltinAudioEncoderFactory.
+  static constexpr const char* GetPayloadName() { return "multiopus"; }
+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format);
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  static absl::optional<AudioEncoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+  static AudioCodecInfo QueryAudioEncoder(
+      const AudioEncoderMultiChannelOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderMultiChannelOpusConfig&,
+      int payload_type);
+
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
+  bool RecreateEncoderInstance(
+      const AudioEncoderMultiChannelOpusConfig& config);
+  void SetFrameLength(int frame_length_ms);
+  void SetNumChannelsToEncode(size_t num_channels_to_encode);
+  void SetProjectedPacketLossRate(float fraction);
+
+  AudioEncoderMultiChannelOpusConfig config_;
+  const int payload_type_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  size_t num_channels_to_encode_;
+  int next_frame_length_ms_;
+
+  friend struct AudioEncoderMultiChannelOpus;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_