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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.h')
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1 files changed, 234 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.h b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.h new file mode 100644 index 0000000000..a61247627f --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.h @@ -0,0 +1,234 @@ +/* + * 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. + */ + +#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ +#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ + +#include <stdint.h> + +#include <map> +#include <memory> +#include <string> +#include <utility> +#include <vector> +#include <atomic> + +#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 { + public: + // Constructor of the class + explicit AcmReceiver(const AudioCodingModule::Config& config); + + // Destructor of the class. + ~AcmReceiver(); + + // + // 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; + + private: + 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_; + + // After construction, this is only ever touched on the thread that calls + // AcmReceiver::GetAudio, and only modified in this method. + std::unique_ptr<int16_t[]> last_audio_buffer_; + CallStatistics call_stats_; + const std::unique_ptr<NetEq> neteq_; // NetEq is thread-safe; no lock needed. + Clock* const clock_; + std::atomic<bool> resampled_last_output_frame_; +}; + +} // namespace acm2 + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ |