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* Copyright (c) 2013 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 <stdint.h>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class Clock;
struct CodecInst;
class NetEq;
struct RTPHeader;
struct WebRtcRTPHeader;
namespace acm2 {
class AcmReceiver {
// Constructor of the class
explicit AcmReceiver(const AudioCodingModule::Config& config);
// Destructor of the class.
// Inserts a payload with its associated RTP-header into NetEq.
// Input:
// - rtp_header : RTP header for the incoming payload containing
// information about payload type, sequence number,
// timestamp, SSRC and marker bit.
// - incoming_payload : Incoming audio payload.
// - length_payload : Length of incoming audio payload in bytes.
// Return value : 0 if OK.
// <0 if NetEq returned an error.
int InsertPacket(const WebRtcRTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> incoming_payload);
// Asks NetEq for 10 milliseconds of decoded audio.
// Input:
// -desired_freq_hz : specifies the sampling rate [Hz] of the output
// audio. If set -1 indicates to resampling is
// is required and the audio returned at the
// sampling rate of the decoder.
// Output:
// -audio_frame : an audio frame were output data and
// associated parameters are written to.
// -muted : if true, the sample data in audio_frame is not
// populated, and must be interpreted as all zero.
// Return value : 0 if OK.
// -1 if NetEq returned an error.
int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
// Replace the current set of decoders with the specified set.
void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
// Adds a new codec to the NetEq codec database.
// Input:
// - acm_codec_id : ACM codec ID; -1 means external decoder.
// - payload_type : payload type.
// - sample_rate_hz : sample rate.
// - audio_decoder : pointer to a decoder object. If it's null, then
// NetEq will internally create a decoder object
// based on the value of |acm_codec_id| (which
// mustn't be -1). Otherwise, NetEq will use the
// given decoder for the given payload type. NetEq
// won't take ownership of the decoder; it's up to
// the caller to delete it when it's no longer
// needed.
// Providing an existing decoder object here is
// necessary for external decoders, but may also be
// used for built-in decoders if NetEq doesn't have
// all the info it needs to construct them properly
// (e.g. iSAC, where the decoder needs to be paired
// with an encoder).
// Return value : 0 if OK.
// <0 if NetEq returned an error.
int AddCodec(int acm_codec_id,
uint8_t payload_type,
size_t channels,
int sample_rate_hz,
AudioDecoder* audio_decoder,
const std::string& name);
// Adds a new decoder to the NetEq codec database. Returns true iff
// successful.
bool AddCodec(int rtp_payload_type, const SdpAudioFormat& audio_format);
// Sets a minimum delay for packet buffer. The given delay is maintained,
// unless channel condition dictates a higher delay.
// Input:
// - delay_ms : minimum delay in milliseconds.
// Return value : 0 if OK.
// <0 if NetEq returned an error.
int SetMinimumDelay(int delay_ms);
// Sets a maximum delay [ms] for the packet buffer. The target delay does not
// exceed the given value, even if channel condition requires so.
// Input:
// - delay_ms : maximum delay in milliseconds.
// Return value : 0 if OK.
// <0 if NetEq returned an error.
int SetMaximumDelay(int delay_ms);
// Resets the initial delay to zero.
void ResetInitialDelay();
// Returns the sample rate of the decoder associated with the last incoming
// packet. If no packet of a registered non-CNG codec has been received, the
// return value is empty. Also, if the decoder was unregistered since the last
// packet was inserted, the return value is empty.
absl::optional<int> last_packet_sample_rate_hz() const;
// Returns last_output_sample_rate_hz from the NetEq instance.
int last_output_sample_rate_hz() const;
// Get the current network statistics from NetEq.
// Output:
// - statistics : The current network statistics.
void GetNetworkStatistics(NetworkStatistics* statistics);
// Flushes the NetEq packet and speech buffers.
void FlushBuffers();
// Removes a payload-type from the NetEq codec database.
// Input:
// - payload_type : the payload-type to be removed.
// Return value : 0 if OK.
// -1 if an error occurred.
int RemoveCodec(uint8_t payload_type);
// Remove all registered codecs.
void RemoveAllCodecs();
// Returns the RTP timestamp for the last sample delivered by GetAudio().
// The return value will be empty if no valid timestamp is available.
absl::optional<uint32_t> GetPlayoutTimestamp();
// Returns the current total delay from NetEq (packet buffer and sync buffer)
// in ms, with smoothing applied to even out short-time fluctuations due to
// jitter. The packet buffer part of the delay is not updated during DTX/CNG
// periods.
int FilteredCurrentDelayMs() const;
// Returns the current target delay for NetEq in ms.
int TargetDelayMs() const;
// Get the audio codec associated with the last non-CNG/non-DTMF received
// payload. If no non-CNG/non-DTMF packet is received -1 is returned,
// otherwise return 0.
int LastAudioCodec(CodecInst* codec) const;
absl::optional<SdpAudioFormat> LastAudioFormat() const;
// Get a decoder given its registered payload-type.
// Input:
// -payload_type : the payload-type of the codec to be retrieved.
// Output:
// -codec : codec associated with the given payload-type.
// Return value : 0 if succeeded.
// -1 if failed, e.g. given payload-type is not
// registered.
int DecoderByPayloadType(uint8_t payload_type,
CodecInst* codec) const;
absl::optional<SdpAudioFormat> DecoderByPayloadType(int payload_type) const;
// Enable NACK and set the maximum size of the NACK list. If NACK is already
// enabled then the maximum NACK list size is modified accordingly.
// Input:
// -max_nack_list_size : maximum NACK list size
// should be positive (none zero) and less than or
// equal to |Nack::kNackListSizeLimit|
// Return value
// : 0 if succeeded.
// -1 if failed
int EnableNack(size_t max_nack_list_size);
// Disable NACK.
void DisableNack();
// Get a list of packets to be retransmitted.
// Input:
// -round_trip_time_ms : estimate of the round-trip-time (in milliseconds).
// Return value : list of packets to be retransmitted.
std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
// Get statistics of calls to GetAudio().
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
struct Decoder {
int acm_codec_id;
uint8_t payload_type;
// This field is meaningful for codecs where both mono and
// stereo versions are registered under the same ID.
size_t channels;
int sample_rate_hz;
const absl::optional<CodecInst> RtpHeaderToDecoder(
const RTPHeader& rtp_header,
uint8_t first_payload_byte) const
uint32_t NowInTimestamp(int decoder_sampling_rate) const;
rtc::CriticalSection crit_sect_;
absl::optional<CodecInst> last_audio_decoder_ RTC_GUARDED_BY(crit_sect_);
absl::optional<SdpAudioFormat> last_audio_format_ RTC_GUARDED_BY(crit_sect_);
ACMResampler resampler_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(crit_sect_);
CallStatistics call_stats_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<NetEq> neteq_; // NetEq is thread-safe; no lock needed.
const Clock* const clock_;
bool resampled_last_output_frame_ RTC_GUARDED_BY(crit_sect_);
absl::optional<int> last_packet_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
} // namespace acm2
} // namespace webrtc