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Diffstat (limited to 'webrtc/modules/audio_coding/main/acm2/acm_receiver.h')
| -rw-r--r-- | webrtc/modules/audio_coding/main/acm2/acm_receiver.h | 348 |
1 files changed, 0 insertions, 348 deletions
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h deleted file mode 100644 index 4775b8c6d9..0000000000 --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h +++ /dev/null @@ -1,348 +0,0 @@ -/* - * 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_ - -#include <map> -#include <vector> - -#include "webrtc/base/scoped_ptr.h" -#include "webrtc/base/thread_annotations.h" -#include "webrtc/common_audio/vad/include/webrtc_vad.h" -#include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" -#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" -#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h" -#include "webrtc/modules/audio_coding/neteq/include/neteq.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/typedefs.h" - -namespace webrtc { - -struct CodecInst; -class CriticalSectionWrapper; -class NetEq; - -namespace acm2 { - -class AcmReceiver { - public: - struct Decoder { - int acm_codec_id; - uint8_t payload_type; - // This field is meaningful for codecs where both mono and - // stereo versions are registered under the same ID. - int channels; - int sample_rate_hz; - }; - - // 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 WebRtcRTPHeader& rtp_header, - const uint8_t* incoming_payload, - size_t length_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. - // - // Return value : 0 if OK. - // -1 if NetEq returned an error. - // - int GetAudio(int desired_freq_hz, AudioFrame* audio_frame); - - // - // Adds a new codec to the NetEq codec database. - // - // Input: - // - acm_codec_id : ACM codec ID; -1 means external decoder. - // - payload_type : payload type. - // - sample_rate_hz : sample rate. - // - audio_decoder : pointer to a decoder object. If it's null, then - // NetEq will internally create a decoder object - // based on the value of |acm_codec_id| (which - // mustn't be -1). Otherwise, NetEq will use the - // given decoder for the given payload type. NetEq - // won't take ownership of the decoder; it's up to - // the caller to delete it when it's no longer - // needed. - // - // Providing an existing decoder object here is - // necessary for external decoders, but may also be - // used for built-in decoders if NetEq doesn't have - // all the info it needs to construct them properly - // (e.g. iSAC, where the decoder needs to be paired - // with an encoder). - // - // Return value : 0 if OK. - // <0 if NetEq returned an error. - // - int AddCodec(int acm_codec_id, - uint8_t payload_type, - int channels, - int sample_rate_hz, - AudioDecoder* audio_decoder); - - // - // 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); - - // - // Get least required delay computed based on channel conditions. Note that - // this is before applying any user-defined limits (specified by calling - // (SetMinimumDelay() and/or SetMaximumDelay()). - // - int LeastRequiredDelayMs() const; - - // - // Sets an initial delay of |delay_ms| milliseconds. This introduces a playout - // delay. Silence (zero signal) is played out until equivalent of |delay_ms| - // millisecond of audio is buffered. Then, NetEq maintains the delay. - // - // Input: - // - delay_ms : initial delay in milliseconds. - // - // Return value : 0 if OK. - // <0 if NetEq returned an error. - // - int SetInitialDelay(int delay_ms); - - // - // Resets the initial delay to zero. - // - void ResetInitialDelay(); - - // - // Get the current sampling frequency in Hz. - // - // Return value : Sampling frequency in Hz. - // - int current_sample_rate_hz() const; - - // - // Get the current network statistics from NetEq. - // - // Output: - // - statistics : The current network statistics. - // - void GetNetworkStatistics(NetworkStatistics* statistics); - - // - // Enable post-decoding VAD. - // - void EnableVad(); - - // - // Disable post-decoding VAD. - // - void DisableVad(); - - // - // Returns whether post-decoding VAD is enabled (true) or disabled (false). - // - bool vad_enabled() const { return vad_enabled_; } - - // - // Flushes the NetEq packet and speech buffers. - // - void FlushBuffers(); - - // - // Removes a payload-type from the NetEq codec database. - // - // Input: - // - payload_type : the payload-type to be removed. - // - // Return value : 0 if OK. - // -1 if an error occurred. - // - int RemoveCodec(uint8_t payload_type); - - // - // Remove all registered codecs. - // - int RemoveAllCodecs(); - - // - // Set ID. - // - void set_id(int id); // TODO(turajs): can be inline. - - // - // Gets the RTP timestamp of the last sample delivered by GetAudio(). - // Returns true if the RTP timestamp is valid, otherwise false. - // - bool GetPlayoutTimestamp(uint32_t* timestamp); - - // - // Return the index of the codec associated with the last non-CNG/non-DTMF - // received payload. If no non-CNG/non-DTMF payload is received -1 is - // returned. - // - int last_audio_codec_id() const; // TODO(turajs): can be inline. - - // - // Get the audio codec associated with the last non-CNG/non-DTMF received - // payload. If no non-CNG/non-DTMF packet is received -1 is returned, - // otherwise return 0. - // - int LastAudioCodec(CodecInst* codec) const; - - // - // Return payload type of RED if it is registered, otherwise return -1; - // - int RedPayloadType() const; - - // - // Get a decoder given its registered payload-type. - // - // Input: - // -payload_type : the payload-type of the codec to be retrieved. - // - // Output: - // -codec : codec associated with the given payload-type. - // - // Return value : 0 if succeeded. - // -1 if failed, e.g. given payload-type is not - // registered. - // - int DecoderByPayloadType(uint8_t payload_type, - CodecInst* codec) 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. - // - // Input: - // -max_nack_list_size : maximum NACK list size - // should be positive (none zero) and less than or - // equal to |Nack::kNackListSizeLimit| - // Return value - // : 0 if succeeded. - // -1 if failed - // - int EnableNack(size_t max_nack_list_size); - - // Disable NACK. - void DisableNack(); - - // - // Get a list of packets to be retransmitted. - // - // Input: - // -round_trip_time_ms : estimate of the round-trip-time (in milliseconds). - // Return value : list of packets to be retransmitted. - // - std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const; - - // - // Get statistics of calls to GetAudio(). - void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; - - private: - bool GetSilence(int desired_sample_rate_hz, AudioFrame* frame) - EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); - - int GetNumSyncPacketToInsert(uint16_t received_squence_number); - - const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header, - const uint8_t* payload) const - EXCLUSIVE_LOCKS_REQUIRED(crit_sect_); - - uint32_t NowInTimestamp(int decoder_sampling_rate) const; - - void InsertStreamOfSyncPackets(InitialDelayManager::SyncStream* sync_stream); - - rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_; - int id_; // TODO(henrik.lundin) Make const. - const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_); - AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_); - int current_sample_rate_hz_ GUARDED_BY(crit_sect_); - ACMResampler resampler_ GUARDED_BY(crit_sect_); - // Used in GetAudio, declared as member to avoid allocating every 10ms. - // TODO(henrik.lundin) Stack-allocate in GetAudio instead? - rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_); - rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_); - CallStatistics call_stats_ GUARDED_BY(crit_sect_); - NetEq* neteq_; - // Decoders map is keyed by payload type - std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_); - bool vad_enabled_; - Clock* clock_; // TODO(henrik.lundin) Make const if possible. - bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); - - // Indicates if a non-zero initial delay is set, and the receiver is in - // AV-sync mode. - bool av_sync_; - rtc::scoped_ptr<InitialDelayManager> initial_delay_manager_; - - // The following are defined as members to avoid creating them in every - // iteration. |missing_packets_sync_stream_| is *ONLY* used in InsertPacket(). - // |late_packets_sync_stream_| is only used in GetAudio(). Both of these - // member variables are allocated only when we AV-sync is enabled, i.e. - // initial delay is set. - rtc::scoped_ptr<InitialDelayManager::SyncStream> missing_packets_sync_stream_; - rtc::scoped_ptr<InitialDelayManager::SyncStream> late_packets_sync_stream_; -}; - -} // namespace acm2 - -} // namespace webrtc - -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_ |