blob: 19dc57758ee46e78e7d0c83b1234f45d91a250cb [file] [log] [blame]
* 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 <utility>
#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/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class Clock;
class NetEq;
struct RTPHeader;
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 RTPHeader& 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);
// 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);
// Sets a base minimum delay in milliseconds for the packet buffer.
// Base minimum delay sets lower bound minimum delay value which
// is set via SetMinimumDelay.
// Returns true if value was successfully set, false overwise.
bool SetBaseMinimumDelayMs(int delay_ms);
// Returns current value of base minimum delay in milliseconds.
int GetBaseMinimumDelayMs() const;
// 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,
bool get_and_clear_legacy_stats = true) const;
// Flushes the NetEq packet and speech buffers.
void FlushBuffers();
// 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 payload type and format of the last non-CNG/non-DTMF received payload.
// If no non-CNG/non-DTMF packet is received absl::nullopt is returned.
absl::optional<std::pair<int, SdpAudioFormat>> LastDecoder() 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.
// If the sequence number of last received packet is N, the sequence numbers
// of NACK list are in the range of [N - |max_nack_list_size|, N).
// |max_nack_list_size| should be positive (none zero) and less than or
// equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
// is returned. 0 is returned at success.
int EnableNack(size_t max_nack_list_size);
// Disable NACK.
void DisableNack();
// Get a list of packets to be retransmitted. |round_trip_time_ms| is an
// estimate of the round-trip-time (in milliseconds). Missing packets which
// will be playout in a shorter time than the round-trip-time (with respect
// to the time this API is called) will not be included in the list.
// Negative |round_trip_time_ms| results is an error message and empty list
// is returned.
std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
// Get statistics of calls to GetAudio().
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
struct DecoderInfo {
int payload_type;
int sample_rate_hz;
int num_channels;
SdpAudioFormat sdp_format;
uint32_t NowInTimestamp(int decoder_sampling_rate) const;
mutable Mutex mutex_;
absl::optional<DecoderInfo> last_decoder_ RTC_GUARDED_BY(mutex_);
ACMResampler resampler_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(mutex_);
CallStatistics call_stats_ RTC_GUARDED_BY(mutex_);
const std::unique_ptr<NetEq> neteq_; // NetEq is thread-safe; no lock needed.
Clock* const clock_;
bool resampled_last_output_frame_ RTC_GUARDED_BY(mutex_);
} // namespace acm2
} // namespace webrtc