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/*
* 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/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"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
class RtcEventLog;
class AudioEncoderOpusImpl final : public AudioEncoder {
public:
class NewPacketLossRateOptimizer {
public:
NewPacketLossRateOptimizer(float min_packet_loss_rate = 0.01,
float max_packet_loss_rate = 0.2,
float slope = 1.0);
float OptimizePacketLossRate(float packet_loss_rate) const;
// Getters for testing.
float min_packet_loss_rate() const { return min_packet_loss_rate_; }
float max_packet_loss_rate() const { return max_packet_loss_rate_; }
float slope() const { return slope_; }
private:
const float min_packet_loss_rate_;
const float max_packet_loss_rate_;
const float slope_;
RTC_DISALLOW_COPY_AND_ASSIGN(NewPacketLossRateOptimizer);
};
// 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>(const std::string&,
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;
// Static interface for use by BuiltinAudioEncoderFactory.
static constexpr const char* GetPayloadName() { return "opus"; }
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;
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 OnReceivedUplinkRecoverablePacketLossFraction(
float uplink_recoverable_packet_loss_fraction) 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;
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_; }
NewPacketLossRateOptimizer* new_packet_loss_optimizer() const {
return new_packet_loss_optimizer_.get();
}
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(
const std::string& config_string,
RtcEventLog* event_log) const;
void MaybeUpdateUplinkBandwidth();
AudioEncoderOpusConfig config_;
const int payload_type_;
const bool send_side_bwe_with_overhead_;
const bool use_link_capacity_for_adaptation_;
const bool adjust_bandwidth_;
bool bitrate_changed_;
float packet_loss_rate_;
const float min_packet_loss_rate_;
const std::unique_ptr<NewPacketLossRateOptimizer> new_packet_loss_optimizer_;
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_;
absl::optional<int64_t> link_capacity_allocation_bps_;
int consecutive_dtx_frames_;
friend struct AudioEncoderOpus;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpusImpl);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
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