blob: 36990fb6708971d2ec37325fc0b6c70809f479f2 [file] [log] [blame]
* Copyright (c) 2012 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 <memory>
#include <string>
#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/defines.h" // Modes, Operations
#include "modules/audio_coding/neteq/expand_uma_logger.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/packet.h"
#include "modules/audio_coding/neteq/random_vector.h"
#include "modules/audio_coding/neteq/rtcp.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
// Forward declarations.
class Accelerate;
class BackgroundNoise;
class BufferLevelFilter;
class ComfortNoise;
class DecisionLogic;
class DecoderDatabase;
class DelayManager;
class DelayPeakDetector;
class DtmfBuffer;
class DtmfToneGenerator;
class Expand;
class Merge;
class NackTracker;
class Normal;
class PacketBuffer;
class RedPayloadSplitter;
class PostDecodeVad;
class PreemptiveExpand;
class RandomVector;
class SyncBuffer;
class TimestampScaler;
struct AccelerateFactory;
struct DtmfEvent;
struct ExpandFactory;
struct PreemptiveExpandFactory;
class NetEqImpl : public webrtc::NetEq {
enum class OutputType {
enum ErrorCodes {
kNoError = 0,
struct Dependencies {
// The constructor populates the Dependencies struct with the default
// implementations of the objects. They can all be replaced by the user
// before sending the struct to the NetEqImpl constructor. However, there
// are dependencies between some of the classes inside the struct, so
// swapping out one may make it necessary to re-create another one.
explicit Dependencies(
const NetEq::Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
std::unique_ptr<TickTimer> tick_timer;
std::unique_ptr<BufferLevelFilter> buffer_level_filter;
std::unique_ptr<DecoderDatabase> decoder_database;
std::unique_ptr<DelayPeakDetector> delay_peak_detector;
std::unique_ptr<DelayManager> delay_manager;
std::unique_ptr<DtmfBuffer> dtmf_buffer;
std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
std::unique_ptr<PacketBuffer> packet_buffer;
std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
std::unique_ptr<TimestampScaler> timestamp_scaler;
std::unique_ptr<AccelerateFactory> accelerate_factory;
std::unique_ptr<ExpandFactory> expand_factory;
std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
// Creates a new NetEqImpl object.
NetEqImpl(const NetEq::Config& config,
Dependencies&& deps,
bool create_components = true);
~NetEqImpl() override;
// Inserts a new packet into NetEq. The |receive_timestamp| is an indication
// of the time when the packet was received, and should be measured with
// the same tick rate as the RTP timestamp of the current payload.
// Returns 0 on success, -1 on failure.
int InsertPacket(const RTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) override;
void InsertEmptyPacket(const RTPHeader& rtp_header) override;
int GetAudio(
AudioFrame* audio_frame,
bool* muted,
absl::optional<Operations> action_override = absl::nullopt) override;
void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
int RegisterPayloadType(NetEqDecoder codec,
const std::string& codec_name,
uint8_t rtp_payload_type) override;
int RegisterExternalDecoder(AudioDecoder* decoder,
NetEqDecoder codec,
const std::string& codec_name,
uint8_t rtp_payload_type) override;
bool RegisterPayloadType(int rtp_payload_type,
const SdpAudioFormat& audio_format) override;
// Removes |rtp_payload_type| from the codec database. Returns 0 on success,
// -1 on failure.
int RemovePayloadType(uint8_t rtp_payload_type) override;
void RemoveAllPayloadTypes() override;
bool SetMinimumDelay(int delay_ms) override;
bool SetMaximumDelay(int delay_ms) override;
int TargetDelayMs() const override;
int CurrentDelayMs() const override;
int FilteredCurrentDelayMs() const override;
// Writes the current network statistics to |stats|. The statistics are reset
// after the call.
int NetworkStatistics(NetEqNetworkStatistics* stats) override;
// Writes the current RTCP statistics to |stats|. The statistics are reset
// and a new report period is started with the call.
void GetRtcpStatistics(RtcpStatistics* stats) override;
NetEqLifetimeStatistics GetLifetimeStatistics() const override;
NetEqOperationsAndState GetOperationsAndState() const override;
// Same as RtcpStatistics(), but does not reset anything.
void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
// Enables post-decode VAD. When enabled, GetAudio() will return
// kOutputVADPassive when the signal contains no speech.
void EnableVad() override;
// Disables post-decode VAD.
void DisableVad() override;
absl::optional<uint32_t> GetPlayoutTimestamp() const override;
int last_output_sample_rate_hz() const override;
absl::optional<CodecInst> GetDecoder(int payload_type) const override;
absl::optional<SdpAudioFormat> GetDecoderFormat(
int payload_type) const override;
// Flushes both the packet buffer and the sync buffer.
void FlushBuffers() override;
void PacketBufferStatistics(int* current_num_packets,
int* max_num_packets) const override;
void EnableNack(size_t max_nack_list_size) override;
void DisableNack() override;
std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
std::vector<uint32_t> LastDecodedTimestamps() const override;
int SyncBufferSizeMs() const override;
// This accessor method is only intended for testing purposes.
const SyncBuffer* sync_buffer_for_test() const;
Operations last_operation_for_test() const;
static const int kOutputSizeMs = 10;
static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
// TODO(hlundin): Provide a better value for kSyncBufferSize.
// Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
// calculating correlations of current frame against history.
static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
// Inserts a new packet into NetEq. This is used by the InsertPacket method
// above. Returns 0 on success, otherwise an error code.
// TODO(hlundin): Merge this with InsertPacket above?
int InsertPacketInternal(const RTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp)
// Delivers 10 ms of audio data. The data is written to |audio_frame|.
// Returns 0 on success, otherwise an error code.
int GetAudioInternal(AudioFrame* audio_frame,
bool* muted,
absl::optional<Operations> action_override)
// Provides a decision to the GetAudioInternal method. The decision what to
// do is written to |operation|. Packets to decode are written to
// |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
// DTMF should be played, |play_dtmf| is set to true by the method.
// Returns 0 on success, otherwise an error code.
int GetDecision(Operations* operation,
PacketList* packet_list,
DtmfEvent* dtmf_event,
bool* play_dtmf,
absl::optional<Operations> action_override)
// Decodes the speech packets in |packet_list|, and writes the results to
// |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
// elements. The length of the decoded data is written to |decoded_length|.
// The speech type -- speech or (codec-internal) comfort noise -- is written
// to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
// comfort noise, those are not decoded.
int Decode(PacketList* packet_list,
Operations* operation,
int* decoded_length,
AudioDecoder::SpeechType* speech_type)
// Sub-method to Decode(). Performs codec internal CNG.
int DecodeCng(AudioDecoder* decoder,
int* decoded_length,
AudioDecoder::SpeechType* speech_type)
// Sub-method to Decode(). Performs the actual decoding.
int DecodeLoop(PacketList* packet_list,
const Operations& operation,
AudioDecoder* decoder,
int* decoded_length,
AudioDecoder::SpeechType* speech_type)
// Sub-method which calls the Normal class to perform the normal operation.
void DoNormal(const int16_t* decoded_buffer,
size_t decoded_length,
AudioDecoder::SpeechType speech_type,
bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Sub-method which calls the Merge class to perform the merge operation.
void DoMerge(int16_t* decoded_buffer,
size_t decoded_length,
AudioDecoder::SpeechType speech_type,
bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
bool DoCodecPlc() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Sub-method which calls the Expand class to perform the expand operation.
int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Sub-method which calls the Accelerate class to perform the accelerate
// operation.
int DoAccelerate(int16_t* decoded_buffer,
size_t decoded_length,
AudioDecoder::SpeechType speech_type,
bool play_dtmf,
bool fast_accelerate)
// Sub-method which calls the PreemptiveExpand class to perform the
// preemtive expand operation.
int DoPreemptiveExpand(int16_t* decoded_buffer,
size_t decoded_length,
AudioDecoder::SpeechType speech_type,
bool play_dtmf)
// Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
// noise. |packet_list| can either contain one SID frame to update the
// noise parameters, or no payload at all, in which case the previously
// received parameters are used.
int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
// Calls the audio decoder to generate codec-internal comfort noise when
// no packet was received.
void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
// Calls the DtmfToneGenerator class to generate DTMF tones.
int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
// Overdub DTMF on top of |output|.
int DtmfOverdub(const DtmfEvent& dtmf_event,
size_t num_channels,
int16_t* output) const
// Extracts packets from |packet_buffer_| to produce at least
// |required_samples| samples. The packets are inserted into |packet_list|.
// Returns the number of samples that the packets in the list will produce, or
// -1 in case of an error.
int ExtractPackets(size_t required_samples, PacketList* packet_list)
// Resets various variables and objects to new values based on the sample rate
// |fs_hz| and |channels| number audio channels.
void SetSampleRateAndChannels(int fs_hz, size_t channels)
// Returns the output type for the audio produced by the latest call to
// GetAudio().
OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Updates Expand and Merge.
virtual void UpdatePlcComponents(int fs_hz, size_t channels)
// Creates DecisionLogic object with the mode given by |playout_mode_|.
virtual void CreateDecisionLogic() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
rtc::CriticalSection crit_sect_;
const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
const std::unique_ptr<DecoderDatabase> decoder_database_
const std::unique_ptr<DelayManager> delay_manager_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
const std::unique_ptr<TimestampScaler> timestamp_scaler_
const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<ExpandFactory> expand_factory_
const std::unique_ptr<AccelerateFactory> accelerate_factory_
const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<DecisionLogic> decision_logic_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<AudioMultiVector> algorithm_buffer_
std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<PreemptiveExpand> preemptive_expand_
RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
Rtcp rtcp_ RTC_GUARDED_BY(crit_sect_);
StatisticsCalculator stats_ RTC_GUARDED_BY(crit_sect_);
int fs_hz_ RTC_GUARDED_BY(crit_sect_);
int fs_mult_ RTC_GUARDED_BY(crit_sect_);
int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
size_t output_size_samples_ RTC_GUARDED_BY(crit_sect_);
size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
Modes last_mode_ RTC_GUARDED_BY(crit_sect_);
Operations last_operation_ RTC_GUARDED_BY(crit_sect_);
size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
bool new_codec_ RTC_GUARDED_BY(crit_sect_);
uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
absl::optional<uint8_t> current_cng_rtp_payload_type_
uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
bool first_packet_ RTC_GUARDED_BY(crit_sect_);
bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_);
bool nack_enabled_ RTC_GUARDED_BY(crit_sect_);
const bool enable_muted_state_ RTC_GUARDED_BY(crit_sect_);
AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(crit_sect_) =
std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
bool no_time_stretching_ RTC_GUARDED_BY(crit_sect_); // Only used for test.
rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(crit_sect_);
} // namespace webrtc