| Index: webrtc/modules/audio_coding/main/acm2/acm_receiver.h
|
| 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 bcedacd14f42a94a7a7c2705c49be0334bb20c55..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
|
| +++ /dev/null
|
| @@ -1,305 +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/array_view.h"
|
| -#include "webrtc/base/optional.h"
|
| -#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_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/include/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,
|
| - 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.
|
| - //
|
| - // 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;
|
| -
|
| - //
|
| - // 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.
|
| - rtc::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);
|
| -
|
| - //
|
| - // 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);
|
| -
|
| - //
|
| - // 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;
|
| -
|
| - //
|
| - // 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:
|
| - const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
|
| - uint8_t payload_type) const
|
| - EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
|
| -
|
| - uint32_t NowInTimestamp(int decoder_sampling_rate) const;
|
| -
|
| - 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_);
|
| - 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_);
|
| - rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
|
| -};
|
| -
|
| -} // namespace acm2
|
| -
|
| -} // namespace webrtc
|
| -
|
| -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
|
|
|