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/*
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* Copyright (c) 2012 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_INCLUDE_AUDIO_CODING_MODULE_H_
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#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
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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/audio_codecs/audio_encoder.h"
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#include "api/function_view.h"
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#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
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namespace webrtc {
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// forward declarations
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class AudioDecoder;
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class AudioEncoder;
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class AudioFrame;
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struct RTPHeader;
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// Callback class used for sending data ready to be packetized
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class AudioPacketizationCallback {
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public:
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virtual ~AudioPacketizationCallback() {}
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virtual int32_t SendData(AudioFrameType frame_type,
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uint8_t payload_type,
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uint32_t timestamp,
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const uint8_t* payload_data,
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size_t payload_len_bytes,
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int64_t absolute_capture_timestamp_ms) {
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// TODO(bugs.webrtc.org/10739): Deprecate the old SendData and make this one
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// pure virtual.
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return SendData(frame_type, payload_type, timestamp, payload_data,
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payload_len_bytes);
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}
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virtual int32_t SendData(AudioFrameType frame_type,
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uint8_t payload_type,
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uint32_t timestamp,
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const uint8_t* payload_data,
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size_t payload_len_bytes) {
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RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
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return -1;
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}
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};
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class AudioCodingModule {
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protected:
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AudioCodingModule() {}
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public:
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static std::unique_ptr<AudioCodingModule> Create();
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virtual ~AudioCodingModule() = default;
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// `modifier` is called exactly once with one argument: a pointer to the
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// unique_ptr that holds the current encoder (which is null if there is no
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// current encoder). For the duration of the call, `modifier` has exclusive
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// access to the unique_ptr; it may call the encoder, steal the encoder and
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// replace it with another encoder or with nullptr, etc.
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virtual void ModifyEncoder(
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rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
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// Utility method for simply replacing the existing encoder with a new one.
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void SetEncoder(std::unique_ptr<AudioEncoder> new_encoder) {
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ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
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*encoder = std::move(new_encoder);
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});
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}
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// int32_t RegisterTransportCallback()
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// Register a transport callback which will be called to deliver
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// the encoded buffers whenever Process() is called and a
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// bit-stream is ready.
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//
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// Input:
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// -transport : pointer to the callback class
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// transport->SendData() is called whenever
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// Process() is called and bit-stream is ready
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// to deliver.
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//
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// Return value:
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// -1 if the transport callback could not be registered
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// 0 if registration is successful.
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//
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virtual int32_t RegisterTransportCallback(
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AudioPacketizationCallback* transport) = 0;
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///////////////////////////////////////////////////////////////////////////
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// int32_t Add10MsData()
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// Add 10MS of raw (PCM) audio data and encode it. If the sampling
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// frequency of the audio does not match the sampling frequency of the
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// current encoder ACM will resample the audio. If an encoded packet was
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// produced, it will be delivered via the callback object registered using
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// RegisterTransportCallback, and the return value from this function will
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// be the number of bytes encoded.
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//
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// Input:
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// -audio_frame : the input audio frame, containing raw audio
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// sampling frequency etc.
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//
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// Return value:
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// >= 0 number of bytes encoded.
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// -1 some error occurred.
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//
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virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0;
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///////////////////////////////////////////////////////////////////////////
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// int SetPacketLossRate()
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// Sets expected packet loss rate for encoding. Some encoders provide packet
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// loss gnostic encoding to make stream less sensitive to packet losses,
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// through e.g., FEC. No effects on codecs that do not provide such encoding.
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//
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// Input:
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// -packet_loss_rate : expected packet loss rate (0 -- 100 inclusive).
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//
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// Return value
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// -1 if failed to set packet loss rate,
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// 0 if succeeded.
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//
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// This is only used in test code that rely on old ACM APIs.
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// TODO(minyue): Remove it when possible.
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virtual int SetPacketLossRate(int packet_loss_rate) = 0;
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///////////////////////////////////////////////////////////////////////////
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// statistics
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//
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virtual ANAStats GetANAStats() const = 0;
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virtual int GetTargetBitrate() const = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
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/*
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* Copyright (c) 2011 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_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
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#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
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#include <map>
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namespace webrtc {
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///////////////////////////////////////////////////////////////////////////
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// enum ACMVADMode
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// An enumerator for aggressiveness of VAD
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// -VADNormal : least aggressive mode.
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// -VADLowBitrate : more aggressive than "VADNormal" to save on
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// bit-rate.
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// -VADAggr : an aggressive mode.
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// -VADVeryAggr : the most agressive mode.
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//
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enum ACMVADMode {
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VADNormal = 0,
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VADLowBitrate = 1,
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VADAggr = 2,
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VADVeryAggr = 3
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};
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enum class AudioFrameType {
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kEmptyFrame = 0,
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kAudioFrameSpeech = 1,
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kAudioFrameCN = 2,
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};
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///////////////////////////////////////////////////////////////////////////
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//
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// Enumeration of Opus mode for intended application.
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//
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// kVoip : optimized for voice signals.
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// kAudio : optimized for non-voice signals like music.
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//
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enum OpusApplicationMode {
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kVoip = 0,
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kAudio = 1,
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};
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// Statistics for calls to AudioCodingModule::PlayoutData10Ms().
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struct AudioDecodingCallStats {
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AudioDecodingCallStats()
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: calls_to_silence_generator(0),
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calls_to_neteq(0),
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decoded_normal(0),
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decoded_neteq_plc(0),
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decoded_codec_plc(0),
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decoded_cng(0),
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decoded_plc_cng(0),
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decoded_muted_output(0) {}
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int calls_to_silence_generator; // Number of calls where silence generated,
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// and NetEq was disengaged from decoding.
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int calls_to_neteq; // Number of calls to NetEq.
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int decoded_normal; // Number of calls where audio RTP packet decoded.
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int decoded_neteq_plc; // Number of calls resulted in NetEq PLC.
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int decoded_codec_plc; // Number of calls resulted in codec PLC.
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int decoded_cng; // Number of calls where comfort noise generated due to DTX.
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int decoded_plc_cng; // Number of calls resulted where PLC faded to CNG.
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int decoded_muted_output; // Number of calls returning a muted state output.
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};
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// NETEQ statistics.
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struct NetworkStatistics {
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// current jitter buffer size in ms
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uint16_t currentBufferSize;
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// preferred (optimal) buffer size in ms
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uint16_t preferredBufferSize;
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// adding extra delay due to "peaky jitter"
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bool jitterPeaksFound;
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// Stats below correspond to similarly-named fields in the WebRTC stats spec.
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// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
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uint64_t totalSamplesReceived;
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uint64_t concealedSamples;
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uint64_t silentConcealedSamples;
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uint64_t concealmentEvents;
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uint64_t jitterBufferDelayMs;
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uint64_t jitterBufferTargetDelayMs;
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uint64_t jitterBufferMinimumDelayMs;
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uint64_t jitterBufferEmittedCount;
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uint64_t insertedSamplesForDeceleration;
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uint64_t removedSamplesForAcceleration;
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uint64_t fecPacketsReceived;
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uint64_t fecPacketsDiscarded;
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// Stats below correspond to similarly-named fields in the WebRTC stats spec.
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// https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats
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uint64_t packetsDiscarded;
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// Stats below DO NOT correspond directly to anything in the WebRTC stats
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// fraction (of original stream) of synthesized audio inserted through
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// expansion (in Q14)
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uint16_t currentExpandRate;
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// fraction (of original stream) of synthesized speech inserted through
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// expansion (in Q14)
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uint16_t currentSpeechExpandRate;
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// fraction of synthesized speech inserted through pre-emptive expansion
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// (in Q14)
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uint16_t currentPreemptiveRate;
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// fraction of data removed through acceleration (in Q14)
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uint16_t currentAccelerateRate;
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// fraction of data coming from secondary decoding (in Q14)
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uint16_t currentSecondaryDecodedRate;
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// Fraction of secondary data, including FEC and RED, that is discarded (in
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// Q14). Discarding of secondary data can be caused by the reception of the
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// primary data, obsoleting the secondary data. It can also be caused by early
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// or late arrival of secondary data.
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uint16_t currentSecondaryDiscardedRate;
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// average packet waiting time in the jitter buffer (ms)
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int meanWaitingTimeMs;
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// max packet waiting time in the jitter buffer (ms)
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int maxWaitingTimeMs;
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// count of the number of buffer flushes
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uint64_t packetBufferFlushes;
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// number of samples expanded due to delayed packets
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uint64_t delayedPacketOutageSamples;
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// arrival delay of incoming packets
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uint64_t relativePacketArrivalDelayMs;
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// number of audio interruptions
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int32_t interruptionCount;
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// total duration of audio interruptions
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int32_t totalInterruptionDurationMs;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
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