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/*
* 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/defines.h"
#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/audio_coding/neteq/packet.h" // Declare PacketList.
#include "webrtc/modules/audio_coding/neteq/random_vector.h"
#include "webrtc/modules/audio_coding/neteq/rtcp.h"
#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Forward declarations.
class Accelerate;
class BackgroundNoise;
class BufferLevelFilter;
class ComfortNoise;
class CriticalSectionWrapper;
class DecisionLogic;
class DecoderDatabase;
class DelayManager;
class DelayPeakDetector;
class DtmfBuffer;
class DtmfToneGenerator;
class Expand;
class Merge;
class Normal;
class PacketBuffer;
class PayloadSplitter;
class PostDecodeVad;
class PreemptiveExpand;
class RandomVector;
class SyncBuffer;
class TimestampScaler;
struct AccelerateFactory;
struct DtmfEvent;
struct ExpandFactory;
struct PreemptiveExpandFactory;
class NetEqImpl : public webrtc::NetEq {
public:
// Creates a new NetEqImpl object. The object will assume ownership of all
// injected dependencies, and will delete them when done.
NetEqImpl(const NetEq::Config& config,
BufferLevelFilter* buffer_level_filter,
DecoderDatabase* decoder_database,
DelayManager* delay_manager,
DelayPeakDetector* delay_peak_detector,
DtmfBuffer* dtmf_buffer,
DtmfToneGenerator* dtmf_tone_generator,
PacketBuffer* packet_buffer,
PayloadSplitter* payload_splitter,
TimestampScaler* timestamp_scaler,
AccelerateFactory* accelerate_factory,
ExpandFactory* expand_factory,
PreemptiveExpandFactory* preemptive_expand_factory,
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 WebRtcRTPHeader& rtp_header,
const uint8_t* payload,
size_t length_bytes,
uint32_t receive_timestamp) override;
// Inserts a sync-packet into packet queue. Sync-packets are decoded to
// silence and are intended to keep AV-sync intact in an event of long packet
// losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
// might insert sync-packet when they observe that buffer level of NetEq is
// decreasing below a certain threshold, defined by the application.
// Sync-packets should have the same payload type as the last audio payload
// type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
// can be implied by inserting a sync-packet.
// Returns kOk on success, kFail on failure.
int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
uint32_t receive_timestamp) override;
// Instructs NetEq to deliver 10 ms of audio data. The data is written to
// |output_audio|, which can hold (at least) |max_length| elements.
// The number of channels that were written to the output is provided in
// the output variable |num_channels|, and each channel contains
// |samples_per_channel| elements. If more than one channel is written,
// the samples are interleaved.
// The speech type is written to |type|, if |type| is not NULL.
// Returns kOK on success, or kFail in case of an error.
int GetAudio(size_t max_length,
int16_t* output_audio,
size_t* samples_per_channel,
int* num_channels,
NetEqOutputType* type) override;
// Associates |rtp_payload_type| with |codec| and stores the information in
// the codec database. Returns kOK on success, kFail on failure.
int RegisterPayloadType(enum NetEqDecoder codec,
uint8_t rtp_payload_type) override;
// Provides an externally created decoder object |decoder| to insert in the
// decoder database. The decoder implements a decoder of type |codec| and
// associates it with |rtp_payload_type|. The decoder will produce samples
// at the rate |sample_rate_hz|. Returns kOK on success, kFail on failure.
int RegisterExternalDecoder(AudioDecoder* decoder,
enum NetEqDecoder codec,
uint8_t rtp_payload_type,
int sample_rate_hz) override;
// Removes |rtp_payload_type| from the codec database. Returns 0 on success,
// -1 on failure.
int RemovePayloadType(uint8_t rtp_payload_type) override;
bool SetMinimumDelay(int delay_ms) override;
bool SetMaximumDelay(int delay_ms) override;
int LeastRequiredDelayMs() const override;
int SetTargetDelay() override;
int TargetDelay() override;
int CurrentDelayMs() const override;
// Sets the playout mode to |mode|.
// Deprecated.
// TODO(henrik.lundin) Delete.
void SetPlayoutMode(NetEqPlayoutMode mode) override;
// Returns the current playout mode.
// Deprecated.
// TODO(henrik.lundin) Delete.
NetEqPlayoutMode PlayoutMode() 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;
// 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;
bool GetPlayoutTimestamp(uint32_t* timestamp) override;
int SetTargetNumberOfChannels() override;
int SetTargetSampleRate() override;
// Returns the error code for the last occurred error. If no error has
// occurred, 0 is returned.
int LastError() const override;
// Returns the error code last returned by a decoder (audio or comfort noise).
// When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
// this method to get the decoder's error code.
int LastDecoderError() 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;
// Get sequence number and timestamp of the latest RTP.
// This method is to facilitate NACK.
int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const override;
// This accessor method is only intended for testing purposes.
const SyncBuffer* sync_buffer_for_test() const;
protected:
static const int kOutputSizeMs = 10;
static const size_t kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
// TODO(hlundin): Provide a better value for kSyncBufferSize.
static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
// 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 WebRtcRTPHeader& rtp_header,
const uint8_t* payload,
size_t length_bytes,
uint32_t receive_timestamp,
bool is_sync_packet)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Delivers 10 ms of audio data. The data is written to |output|, which can
// hold (at least) |max_length| elements. The number of channels that were
// written to the output is provided in the output variable |num_channels|,
// and each channel contains |samples_per_channel| elements. If more than one
// channel is written, the samples are interleaved.
// Returns 0 on success, otherwise an error code.
int GetAudioInternal(size_t max_length,
int16_t* output,
size_t* samples_per_channel,
int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// 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) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// 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)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Sub-method to Decode(). Performs the actual decoding.
int DecodeLoop(PacketList* packet_list,
Operations* operation,
AudioDecoder* decoder,
int* decoded_length,
AudioDecoder::SpeechType* speech_type)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// 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) 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) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Sub-method which calls the Expand class to perform the expand operation.
int DoExpand(bool play_dtmf) 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) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// 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) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// 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)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Calls the audio decoder to generate codec-internal comfort noise when
// no packet was received.
void DoCodecInternalCng() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Calls the DtmfToneGenerator class to generate DTMF tones.
int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Produces packet-loss concealment using alternative methods. If the codec
// has an internal PLC, it is called to generate samples. Otherwise, the
// method performs zero-stuffing.
void DoAlternativePlc(bool increase_timestamp)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Overdub DTMF on top of |output|.
int DtmfOverdub(const DtmfEvent& dtmf_event,
size_t num_channels,
int16_t* output) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// 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)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// 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)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Returns the output type for the audio produced by the latest call to
// GetAudio().
NetEqOutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Updates Expand and Merge.
virtual void UpdatePlcComponents(int fs_hz, size_t channels)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Creates DecisionLogic object with the mode given by |playout_mode_|.
virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
const rtc::scoped_ptr<BufferLevelFilter> buffer_level_filter_
GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<DecoderDatabase> decoder_database_
GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<DelayPeakDetector> delay_peak_detector_
GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<DtmfToneGenerator> dtmf_tone_generator_
GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<PayloadSplitter> payload_splitter_
GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<TimestampScaler> timestamp_scaler_
GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<AccelerateFactory> accelerate_factory_
GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
GUARDED_BY(crit_sect_);
rtc::scoped_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
RandomVector random_vector_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
Rtcp rtcp_ GUARDED_BY(crit_sect_);
StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
int fs_hz_ GUARDED_BY(crit_sect_);
int fs_mult_ GUARDED_BY(crit_sect_);
size_t output_size_samples_ GUARDED_BY(crit_sect_);
size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
Modes last_mode_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
bool new_codec_ GUARDED_BY(crit_sect_);
uint32_t timestamp_ GUARDED_BY(crit_sect_);
bool reset_decoder_ GUARDED_BY(crit_sect_);
uint8_t current_rtp_payload_type_ GUARDED_BY(crit_sect_);
uint8_t current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_);
uint32_t ssrc_ GUARDED_BY(crit_sect_);
bool first_packet_ GUARDED_BY(crit_sect_);
int error_code_ GUARDED_BY(crit_sect_); // Store last error code.
int decoder_error_code_ GUARDED_BY(crit_sect_);
const BackgroundNoiseMode background_noise_mode_ GUARDED_BY(crit_sect_);
NetEqPlayoutMode playout_mode_ GUARDED_BY(crit_sect_);
bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
// These values are used by NACK module to estimate time-to-play of
// a missing packet. Occasionally, NetEq might decide to decode more
// than one packet. Therefore, these values store sequence number and
// timestamp of the first packet pulled from the packet buffer. In
// such cases, these values do not exactly represent the sequence number
// or timestamp associated with a 10ms audio pulled from NetEq. NACK
// module is designed to compensate for this.
int decoded_packet_sequence_number_ GUARDED_BY(crit_sect_);
uint32_t decoded_packet_timestamp_ GUARDED_BY(crit_sect_);
private:
DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_