230 lines
7.6 KiB
C++
230 lines
7.6 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
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#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
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#include <stdint.h>
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#include <map>
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#include <memory>
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/audio_codecs/audio_decoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "modules/audio_coding/acm2/acm_resampler.h"
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#include "modules/audio_coding/acm2/call_statistics.h"
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class Clock;
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class NetEq;
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struct RTPHeader;
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namespace acm2 {
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class AcmReceiver {
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public:
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// Constructor of the class
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explicit AcmReceiver(const AudioCodingModule::Config& config);
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// Destructor of the class.
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~AcmReceiver();
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//
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// Inserts a payload with its associated RTP-header into NetEq.
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//
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// Input:
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// - rtp_header : RTP header for the incoming payload containing
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// information about payload type, sequence number,
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// timestamp, SSRC and marker bit.
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// - incoming_payload : Incoming audio payload.
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// - length_payload : Length of incoming audio payload in bytes.
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int InsertPacket(const RTPHeader& rtp_header,
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rtc::ArrayView<const uint8_t> incoming_payload);
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//
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// Asks NetEq for 10 milliseconds of decoded audio.
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//
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// Input:
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// -desired_freq_hz : specifies the sampling rate [Hz] of the output
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// audio. If set -1 indicates to resampling is
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// is required and the audio returned at the
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// sampling rate of the decoder.
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//
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// Output:
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// -audio_frame : an audio frame were output data and
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// associated parameters are written to.
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// -muted : if true, the sample data in audio_frame is not
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// populated, and must be interpreted as all zero.
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//
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// Return value : 0 if OK.
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// -1 if NetEq returned an error.
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//
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int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
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// Replace the current set of decoders with the specified set.
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void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
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//
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// Sets a minimum delay for packet buffer. The given delay is maintained,
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// unless channel condition dictates a higher delay.
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//
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// Input:
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// - delay_ms : minimum delay in milliseconds.
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int SetMinimumDelay(int delay_ms);
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//
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// Sets a maximum delay [ms] for the packet buffer. The target delay does not
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// exceed the given value, even if channel condition requires so.
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//
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// Input:
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// - delay_ms : maximum delay in milliseconds.
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int SetMaximumDelay(int delay_ms);
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// Sets a base minimum delay in milliseconds for the packet buffer.
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// Base minimum delay sets lower bound minimum delay value which
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// is set via SetMinimumDelay.
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//
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// Returns true if value was successfully set, false overwise.
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bool SetBaseMinimumDelayMs(int delay_ms);
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// Returns current value of base minimum delay in milliseconds.
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int GetBaseMinimumDelayMs() const;
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//
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// Resets the initial delay to zero.
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//
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void ResetInitialDelay();
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// Returns the sample rate of the decoder associated with the last incoming
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// packet. If no packet of a registered non-CNG codec has been received, the
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// return value is empty. Also, if the decoder was unregistered since the last
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// packet was inserted, the return value is empty.
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absl::optional<int> last_packet_sample_rate_hz() const;
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// Returns last_output_sample_rate_hz from the NetEq instance.
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int last_output_sample_rate_hz() const;
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//
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// Get the current network statistics from NetEq.
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//
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// Output:
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// - statistics : The current network statistics.
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//
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void GetNetworkStatistics(NetworkStatistics* statistics) const;
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//
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// Flushes the NetEq packet and speech buffers.
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//
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void FlushBuffers();
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//
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// Remove all registered codecs.
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//
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void RemoveAllCodecs();
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// Returns the RTP timestamp for the last sample delivered by GetAudio().
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// The return value will be empty if no valid timestamp is available.
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absl::optional<uint32_t> GetPlayoutTimestamp();
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// Returns the current total delay from NetEq (packet buffer and sync buffer)
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// in ms, with smoothing applied to even out short-time fluctuations due to
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// jitter. The packet buffer part of the delay is not updated during DTX/CNG
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// periods.
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//
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int FilteredCurrentDelayMs() const;
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// Returns the current target delay for NetEq in ms.
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//
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int TargetDelayMs() const;
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//
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// Get payload type and format of the last non-CNG/non-DTMF received payload.
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// If no non-CNG/non-DTMF packet is received absl::nullopt is returned.
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//
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absl::optional<std::pair<int, SdpAudioFormat>> LastDecoder() const;
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//
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// Enable NACK and set the maximum size of the NACK list. If NACK is already
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// enabled then the maximum NACK list size is modified accordingly.
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//
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// If the sequence number of last received packet is N, the sequence numbers
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// of NACK list are in the range of [N - |max_nack_list_size|, N).
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//
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// |max_nack_list_size| should be positive (none zero) and less than or
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// equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
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// is returned. 0 is returned at success.
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//
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int EnableNack(size_t max_nack_list_size);
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// Disable NACK.
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void DisableNack();
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//
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// Get a list of packets to be retransmitted. |round_trip_time_ms| is an
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// estimate of the round-trip-time (in milliseconds). Missing packets which
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// will be playout in a shorter time than the round-trip-time (with respect
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// to the time this API is called) will not be included in the list.
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//
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// Negative |round_trip_time_ms| results is an error message and empty list
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// is returned.
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//
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std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
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//
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// Get statistics of calls to GetAudio().
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void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
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private:
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struct DecoderInfo {
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int payload_type;
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int sample_rate_hz;
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int num_channels;
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SdpAudioFormat sdp_format;
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};
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uint32_t NowInTimestamp(int decoder_sampling_rate) const;
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mutable Mutex mutex_;
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absl::optional<DecoderInfo> last_decoder_ RTC_GUARDED_BY(mutex_);
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ACMResampler resampler_ RTC_GUARDED_BY(mutex_);
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std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(mutex_);
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CallStatistics call_stats_ RTC_GUARDED_BY(mutex_);
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const std::unique_ptr<NetEq> neteq_; // NetEq is thread-safe; no lock needed.
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Clock* const clock_;
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bool resampled_last_output_frame_ RTC_GUARDED_BY(mutex_);
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};
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} // namespace acm2
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
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