diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h | 184 |
1 files changed, 184 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h new file mode 100644 index 0000000000..8c5c235016 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h @@ -0,0 +1,184 @@ +/* + * 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 MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ +#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ + +#include <functional> +#include <memory> +#include <string> +#include <vector> + +#include "absl/strings/string_view.h" +#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_opus_config.h" +#include "common_audio/smoothing_filter.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" +#include "modules/audio_coding/codecs/opus/opus_interface.h" + +namespace webrtc { + +class RtcEventLog; + +class AudioEncoderOpusImpl final : public AudioEncoder { + public: + // Returns empty if the current bitrate falls within the hysteresis window, + // defined by complexity_threshold_bps +/- complexity_threshold_window_bps. + // Otherwise, returns the current complexity depending on whether the + // current bitrate is above or below complexity_threshold_bps. + static absl::optional<int> GetNewComplexity( + const AudioEncoderOpusConfig& config); + + // Returns OPUS_AUTO if the the current bitrate is above wideband threshold. + // Returns empty if it is below, but bandwidth coincides with the desired one. + // Otherwise returns the desired bandwidth. + static absl::optional<int> GetNewBandwidth( + const AudioEncoderOpusConfig& config, + OpusEncInst* inst); + + using AudioNetworkAdaptorCreator = + std::function<std::unique_ptr<AudioNetworkAdaptor>(absl::string_view, + RtcEventLog*)>; + + AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config, int payload_type); + + // Dependency injection for testing. + AudioEncoderOpusImpl( + const AudioEncoderOpusConfig& config, + int payload_type, + const AudioNetworkAdaptorCreator& audio_network_adaptor_creator, + std::unique_ptr<SmoothingFilter> bitrate_smoother); + + AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format); + ~AudioEncoderOpusImpl() override; + + AudioEncoderOpusImpl(const AudioEncoderOpusImpl&) = delete; + AudioEncoderOpusImpl& operator=(const AudioEncoderOpusImpl&) = delete; + + int SampleRateHz() const override; + size_t NumChannels() const override; + int RtpTimestampRateHz() const override; + size_t Num10MsFramesInNextPacket() const override; + size_t Max10MsFramesInAPacket() const override; + int GetTargetBitrate() const override; + + void Reset() override; + bool SetFec(bool enable) override; + + // Set Opus DTX. Once enabled, Opus stops transmission, when it detects + // voice being inactive. During that, it still sends 2 packets (one for + // content, one for signaling) about every 400 ms. + bool SetDtx(bool enable) override; + bool GetDtx() const override; + + bool SetApplication(Application application) override; + void SetMaxPlaybackRate(int frequency_hz) override; + bool EnableAudioNetworkAdaptor(const std::string& config_string, + RtcEventLog* event_log) override; + void DisableAudioNetworkAdaptor() override; + void OnReceivedUplinkPacketLossFraction( + float uplink_packet_loss_fraction) override; + void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override; + void OnReceivedUplinkBandwidth( + int target_audio_bitrate_bps, + absl::optional<int64_t> bwe_period_ms) override; + void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override; + void OnReceivedRtt(int rtt_ms) override; + void OnReceivedOverhead(size_t overhead_bytes_per_packet) override; + void SetReceiverFrameLengthRange(int min_frame_length_ms, + int max_frame_length_ms) override; + ANAStats GetANAStats() const override; + absl::optional<std::pair<TimeDelta, TimeDelta> > GetFrameLengthRange() + const override; + rtc::ArrayView<const int> supported_frame_lengths_ms() const { + return config_.supported_frame_lengths_ms; + } + + // Getters for testing. + float packet_loss_rate() const { return packet_loss_rate_; } + AudioEncoderOpusConfig::ApplicationMode application() const { + return config_.application; + } + bool fec_enabled() const { return config_.fec_enabled; } + size_t num_channels_to_encode() const { return num_channels_to_encode_; } + int next_frame_length_ms() const { return next_frame_length_ms_; } + + protected: + EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) override; + + private: + class PacketLossFractionSmoother; + + static absl::optional<AudioEncoderOpusConfig> SdpToConfig( + const SdpAudioFormat& format); + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); + static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config); + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const AudioEncoderOpusConfig&, + int payload_type); + + size_t Num10msFramesPerPacket() const; + size_t SamplesPer10msFrame() const; + size_t SufficientOutputBufferSize() const; + bool RecreateEncoderInstance(const AudioEncoderOpusConfig& config); + void SetFrameLength(int frame_length_ms); + void SetNumChannelsToEncode(size_t num_channels_to_encode); + void SetProjectedPacketLossRate(float fraction); + + void OnReceivedUplinkBandwidth( + int target_audio_bitrate_bps, + absl::optional<int64_t> bwe_period_ms, + absl::optional<int64_t> link_capacity_allocation); + + // TODO(minyue): remove "override" when we can deprecate + // `AudioEncoder::SetTargetBitrate`. + void SetTargetBitrate(int target_bps) override; + + void ApplyAudioNetworkAdaptor(); + std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator( + absl::string_view config_string, + RtcEventLog* event_log) const; + + void MaybeUpdateUplinkBandwidth(); + + AudioEncoderOpusConfig config_; + const int payload_type_; + const bool use_stable_target_for_adaptation_; + const bool adjust_bandwidth_; + bool bitrate_changed_; + // A multiplier for bitrates at 5 kbps and higher. The target bitrate + // will be multiplied by these multipliers, each multiplier is applied to a + // 1 kbps range. + std::vector<float> bitrate_multipliers_; + float packet_loss_rate_; + 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_; + int complexity_; + std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_; + const AudioNetworkAdaptorCreator audio_network_adaptor_creator_; + std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_; + absl::optional<size_t> overhead_bytes_per_packet_; + const std::unique_ptr<SmoothingFilter> bitrate_smoother_; + absl::optional<int64_t> bitrate_smoother_last_update_time_; + int consecutive_dtx_frames_; + + friend struct AudioEncoderOpus; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ |