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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h')
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1 files changed, 246 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h new file mode 100644 index 0000000000..8b518fb979 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h @@ -0,0 +1,246 @@ +/* + * Copyright (c) 2012 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_INCLUDE_AUDIO_CODING_MODULE_H_ +#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ + +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/function_view.h" +#include "api/neteq/neteq.h" +#include "api/neteq/neteq_factory.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "system_wrappers/include/clock.h" + +namespace webrtc { + +// forward declarations +class AudioDecoder; +class AudioEncoder; +class AudioFrame; +struct RTPHeader; + +// Callback class used for sending data ready to be packetized +class AudioPacketizationCallback { + public: + virtual ~AudioPacketizationCallback() {} + + virtual int32_t SendData(AudioFrameType frame_type, + uint8_t payload_type, + uint32_t timestamp, + const uint8_t* payload_data, + size_t payload_len_bytes, + int64_t absolute_capture_timestamp_ms) { + // TODO(bugs.webrtc.org/10739): Deprecate the old SendData and make this one + // pure virtual. + return SendData(frame_type, payload_type, timestamp, payload_data, + payload_len_bytes); + } + virtual int32_t SendData(AudioFrameType frame_type, + uint8_t payload_type, + uint32_t timestamp, + const uint8_t* payload_data, + size_t payload_len_bytes) { + RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used."; + return -1; + } +}; + +class AudioCodingModule { + protected: + AudioCodingModule() {} + + public: + struct Config { + explicit Config( + rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr); + Config(const Config&); + ~Config(); + + NetEq::Config neteq_config; + Clock* clock; + rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; + NetEqFactory* neteq_factory = nullptr; + }; + + static AudioCodingModule* Create(const Config& config); + virtual ~AudioCodingModule() = default; + + /////////////////////////////////////////////////////////////////////////// + // Sender + // + + // `modifier` is called exactly once with one argument: a pointer to the + // unique_ptr that holds the current encoder (which is null if there is no + // current encoder). For the duration of the call, `modifier` has exclusive + // access to the unique_ptr; it may call the encoder, steal the encoder and + // replace it with another encoder or with nullptr, etc. + virtual void ModifyEncoder( + rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0; + + // Utility method for simply replacing the existing encoder with a new one. + void SetEncoder(std::unique_ptr<AudioEncoder> new_encoder) { + ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { + *encoder = std::move(new_encoder); + }); + } + + // int32_t RegisterTransportCallback() + // Register a transport callback which will be called to deliver + // the encoded buffers whenever Process() is called and a + // bit-stream is ready. + // + // Input: + // -transport : pointer to the callback class + // transport->SendData() is called whenever + // Process() is called and bit-stream is ready + // to deliver. + // + // Return value: + // -1 if the transport callback could not be registered + // 0 if registration is successful. + // + virtual int32_t RegisterTransportCallback( + AudioPacketizationCallback* transport) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t Add10MsData() + // Add 10MS of raw (PCM) audio data and encode it. If the sampling + // frequency of the audio does not match the sampling frequency of the + // current encoder ACM will resample the audio. If an encoded packet was + // produced, it will be delivered via the callback object registered using + // RegisterTransportCallback, and the return value from this function will + // be the number of bytes encoded. + // + // Input: + // -audio_frame : the input audio frame, containing raw audio + // sampling frequency etc. + // + // Return value: + // >= 0 number of bytes encoded. + // -1 some error occurred. + // + virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int SetPacketLossRate() + // Sets expected packet loss rate for encoding. Some encoders provide packet + // loss gnostic encoding to make stream less sensitive to packet losses, + // through e.g., FEC. No effects on codecs that do not provide such encoding. + // + // Input: + // -packet_loss_rate : expected packet loss rate (0 -- 100 inclusive). + // + // Return value + // -1 if failed to set packet loss rate, + // 0 if succeeded. + // + // This is only used in test code that rely on old ACM APIs. + // TODO(minyue): Remove it when possible. + virtual int SetPacketLossRate(int packet_loss_rate) = 0; + + /////////////////////////////////////////////////////////////////////////// + // Receiver + // + + /////////////////////////////////////////////////////////////////////////// + // int32_t InitializeReceiver() + // Any decoder-related state of ACM will be initialized to the + // same state when ACM is created. This will not interrupt or + // effect encoding functionality of ACM. ACM would lose all the + // decoding-related settings by calling this function. + // For instance, all registered codecs are deleted and have to be + // registered again. + // + // Return value: + // -1 if failed to initialize, + // 0 if succeeded. + // + virtual int32_t InitializeReceiver() = 0; + + // Replace any existing decoders with the given payload type -> decoder map. + virtual void SetReceiveCodecs( + const std::map<int, SdpAudioFormat>& codecs) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t IncomingPacket() + // Call this function to insert a parsed RTP packet into ACM. + // + // Inputs: + // -incoming_payload : received payload. + // -payload_len_bytes : the length of payload in bytes. + // -rtp_info : the relevant information retrieved from RTP + // header. + // + // Return value: + // -1 if failed to push in the payload + // 0 if payload is successfully pushed in. + // + virtual int32_t IncomingPacket(const uint8_t* incoming_payload, + size_t payload_len_bytes, + const RTPHeader& rtp_header) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t PlayoutData10Ms( + // Get 10 milliseconds of raw audio data for playout, at the given sampling + // frequency. ACM will perform a resampling if required. + // + // Input: + // -desired_freq_hz : the desired sampling frequency, in Hertz, of the + // output audio. If set to -1, the function returns + // the audio at the current sampling frequency. + // + // Output: + // -audio_frame : output audio frame which contains raw audio data + // and other relevant parameters. + // -muted : if true, the sample data in audio_frame is not + // populated, and must be interpreted as all zero. + // + // Return value: + // -1 if the function fails, + // 0 if the function succeeds. + // + virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz, + AudioFrame* audio_frame, + bool* muted) = 0; + + /////////////////////////////////////////////////////////////////////////// + // statistics + // + + /////////////////////////////////////////////////////////////////////////// + // int32_t GetNetworkStatistics() + // Get network statistics. Note that the internal statistics of NetEq are + // reset by this call. + // + // Input: + // -network_statistics : a structure that contains network statistics. + // + // Return value: + // -1 if failed to set the network statistics, + // 0 if statistics are set successfully. + // + virtual int32_t GetNetworkStatistics( + NetworkStatistics* network_statistics) = 0; + + virtual ANAStats GetANAStats() const = 0; + + virtual int GetTargetBitrate() const = 0; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ |