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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h | 92 |
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_ |