Files
platform-external-webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
henrik.lundin b8c55b15a3 Handle padded audio packets correctly
RTP packets can be padded with extra data at the end of the payload. The usable
payload length of the packet should then be reduced with the padding length,
since the padding must be discarded. This was not the case; instead, the entire
payload, including padding data, was forwarded to the audio channel and in the
end to the decoder.

A special case of padding is packets which are empty except for the padding.
That is, they carry no usable payload. These packets are sometimes used for
probing the network and were discarded in
RTPReceiverAudio::ParseAudioCodecSpecific. The result is that NetEq never sees
those empty packets, just the holes in the sequence number series; this can
throw off the target buffer calculations.

With this change, the empty (after removing the padding) packets are let through,
all the way down to NetEq, to a new method called NetEq::InsertEmptyPacket. This
method notifies the DelayManager that an empty packet was received.

BUG=webrtc:7610, webrtc:7625

Review-Url: https://codereview.webrtc.org/2870043003
Cr-Commit-Position: refs/heads/master@{#18083}
2017-05-10 14:38:01 +00:00

431 lines
18 KiB
C++

/*
* 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 <memory>
#include <string>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/optional.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/include/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/modules/audio_coding/neteq/tick_timer.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.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 {
public:
enum class OutputType {
kNormalSpeech,
kPLC,
kCNG,
kPLCCNG,
kVadPassive
};
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);
~Dependencies();
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) 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 LeastRequiredDelayMs() const override;
int SetTargetDelay() override;
int TargetDelayMs() override;
int CurrentDelayMs() const override;
int FilteredCurrentDelayMs() 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;
rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
int last_output_sample_rate_hz() const override;
rtc::Optional<CodecInst> GetDecoder(int payload_type) const override;
rtc::Optional<SdpAudioFormat> GetDecoderFormat(
int payload_type) const 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;
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;
protected:
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)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// 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)
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 codec internal CNG.
int DecodeCng(AudioDecoder* decoder, 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,
const 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(const int16_t* decoded_buffer, size_t decoded_length)
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().
OutputType 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_);
rtc::CriticalSection crit_sect_;
const std::unique_ptr<TickTimer> tick_timer_ GUARDED_BY(crit_sect_);
const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
GUARDED_BY(crit_sect_);
const std::unique_ptr<DecoderDatabase> decoder_database_
GUARDED_BY(crit_sect_);
const std::unique_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
GUARDED_BY(crit_sect_);
const std::unique_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
GUARDED_BY(crit_sect_);
const std::unique_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
GUARDED_BY(crit_sect_);
const std::unique_ptr<TimestampScaler> timestamp_scaler_
GUARDED_BY(crit_sect_);
const std::unique_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
const std::unique_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
const std::unique_ptr<AccelerateFactory> accelerate_factory_
GUARDED_BY(crit_sect_);
const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
GUARDED_BY(crit_sect_);
std::unique_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
std::unique_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
std::unique_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
std::unique_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
std::unique_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
std::unique_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
std::unique_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
std::unique_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
std::unique_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
RandomVector random_vector_ GUARDED_BY(crit_sect_);
std::unique_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_);
int last_output_sample_rate_hz_ 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_);
Operations last_operation_ GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
std::unique_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_);
rtc::Optional<uint8_t> current_rtp_payload_type_ GUARDED_BY(crit_sect_);
rtc::Optional<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_);
std::unique_ptr<NackTracker> nack_ GUARDED_BY(crit_sect_);
bool nack_enabled_ GUARDED_BY(crit_sect_);
const bool enable_muted_state_ GUARDED_BY(crit_sect_);
AudioFrame::VADActivity last_vad_activity_ GUARDED_BY(crit_sect_) =
AudioFrame::kVadPassive;
std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
GUARDED_BY(crit_sect_);
std::vector<uint32_t> last_decoded_timestamps_ GUARDED_BY(crit_sect_);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_