2013-09-12 18:30:26 +00:00
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/*
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* Copyright (c) 2013 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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2017-09-15 06:47:31 +02:00
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#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
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#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
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2013-09-12 18:30:26 +00:00
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2018-10-23 12:03:01 +02:00
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#include <stdint.h>
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2019-07-05 19:08:33 +02:00
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2024-05-29 09:52:55 +02:00
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#include <array>
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2015-03-23 11:19:35 +00:00
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#include <map>
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2016-02-15 02:27:22 -08:00
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#include <memory>
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2024-08-29 13:00:40 +00:00
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#include <optional>
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2015-12-09 06:20:58 -08:00
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#include <string>
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Remove CodecInst pt.2
The following APIs on AudioCodingModule are deprecated with this CL:
static int NumberOfCodecs();
static int Codec(int, CodecInst*);
static int Codec(const char*, CodecInst*, int, size_t);
static int Codec(const char*, int, size_t);
absl::optional<CodecInst> SendCodec() const;
bool RegisterReceiveCodec(int, const SdpAudioFormat&);
int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
int UnregisterReceiveCodec(uint8_t);
int32_t ReceiveCodec(CodecInst*);
absl::optional<SdpAudioFormat> ReceiveFormat();
As well as this method on RtpRtcp module:
int32_t RegisterSendPayload(const CodecInst&);
Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}
2018-12-11 12:22:10 +01:00
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#include <utility>
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#include <vector>
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#include "api/array_view.h"
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#include "api/audio/audio_frame.h"
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#include "api/audio_codecs/audio_decoder.h"
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#include "api/audio_codecs/audio_decoder_factory.h"
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#include "api/audio_codecs/audio_format.h"
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#include "api/environment/environment.h"
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#include "api/neteq/neteq.h"
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#include "api/neteq/neteq_factory.h"
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#include "api/units/timestamp.h"
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#include "modules/audio_coding/acm2/acm_resampler.h"
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#include "modules/audio_coding/acm2/call_statistics.h"
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#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class NetEq;
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struct RTPHeader;
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2013-10-06 04:47:28 +00:00
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namespace acm2 {
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// This class is deprecated. See https://issues.webrtc.org/issues/42225167.
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class AcmReceiver {
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public:
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struct Config {
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explicit Config(
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr);
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Config(const Config&);
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~Config();
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NetEq::Config neteq_config;
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
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NetEqFactory* neteq_factory = nullptr;
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};
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AcmReceiver(const Environment& env, Config config);
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// Destructor of the class.
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~AcmReceiver();
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//
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// Inserts a payload with its associated RTP-header into NetEq.
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//
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// Input:
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// - rtp_header : RTP header for the incoming payload containing
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// information about payload type, sequence number,
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// timestamp, SSRC and marker bit.
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// - incoming_payload : Incoming audio payload.
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// - receive_time : Timestamp when the packet has been seen on the
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// network card.
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int InsertPacket(const RTPHeader& rtp_header,
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rtc::ArrayView<const uint8_t> incoming_payload,
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Timestamp receive_time = Timestamp::MinusInfinity());
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//
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// Asks NetEq for 10 milliseconds of decoded audio.
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//
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// Input:
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// -desired_freq_hz : specifies the sampling rate [Hz] of the output
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// audio. If set -1 indicates to resampling is
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// is required and the audio returned at the
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// sampling rate of the decoder.
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//
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// Output:
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// -audio_frame : an audio frame were output data and
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// associated parameters are written to.
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// -muted : if true, the sample data in audio_frame is not
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// populated, and must be interpreted as all zero.
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//
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// Return value : 0 if OK.
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// -1 if NetEq returned an error.
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//
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int GetAudio(int desired_freq_hz,
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AudioFrame* audio_frame,
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bool* muted = nullptr);
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// Replace the current set of decoders with the specified set.
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void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
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2013-09-12 18:30:26 +00:00
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//
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// Sets a minimum delay for packet buffer. The given delay is maintained,
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// unless channel condition dictates a higher delay.
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//
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// Input:
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// - delay_ms : minimum delay in milliseconds.
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int SetMinimumDelay(int delay_ms);
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//
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// Sets a maximum delay [ms] for the packet buffer. The target delay does not
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// exceed the given value, even if channel condition requires so.
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//
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// Input:
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// - delay_ms : maximum delay in milliseconds.
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int SetMaximumDelay(int delay_ms);
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2019-02-05 13:49:26 +01:00
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// Sets a base minimum delay in milliseconds for the packet buffer.
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// Base minimum delay sets lower bound minimum delay value which
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// is set via SetMinimumDelay.
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//
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// Returns true if value was successfully set, false overwise.
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bool SetBaseMinimumDelayMs(int delay_ms);
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// Returns current value of base minimum delay in milliseconds.
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int GetBaseMinimumDelayMs() const;
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//
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// Resets the initial delay to zero.
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//
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void ResetInitialDelay();
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// Returns the sample rate of the decoder associated with the last incoming
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// packet. If no packet of a registered non-CNG codec has been received, the
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// return value is empty. Also, if the decoder was unregistered since the last
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// packet was inserted, the return value is empty.
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std::optional<int> last_packet_sample_rate_hz() const;
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// Returns last_output_sample_rate_hz from the NetEq instance.
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int last_output_sample_rate_hz() const;
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//
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// Get the current network statistics from NetEq.
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//
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// Output:
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// - statistics : The current network statistics.
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//
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2020-09-14 10:47:50 +02:00
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void GetNetworkStatistics(NetworkStatistics* statistics,
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bool get_and_clear_legacy_stats = true) const;
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//
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// Flushes the NetEq packet and speech buffers.
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//
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void FlushBuffers();
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2016-04-06 01:39:22 -07:00
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// Returns the RTP timestamp for the last sample delivered by GetAudio().
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// The return value will be empty if no valid timestamp is available.
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std::optional<uint32_t> GetPlayoutTimestamp();
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// Returns the current total delay from NetEq (packet buffer and sync buffer)
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// in ms, with smoothing applied to even out short-time fluctuations due to
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// jitter. The packet buffer part of the delay is not updated during DTX/CNG
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// periods.
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//
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int FilteredCurrentDelayMs() const;
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// Returns the current target delay for NetEq in ms.
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//
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int TargetDelayMs() const;
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2013-09-12 18:30:26 +00:00
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//
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Remove CodecInst pt.2
The following APIs on AudioCodingModule are deprecated with this CL:
static int NumberOfCodecs();
static int Codec(int, CodecInst*);
static int Codec(const char*, CodecInst*, int, size_t);
static int Codec(const char*, int, size_t);
absl::optional<CodecInst> SendCodec() const;
bool RegisterReceiveCodec(int, const SdpAudioFormat&);
int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
int UnregisterReceiveCodec(uint8_t);
int32_t ReceiveCodec(CodecInst*);
absl::optional<SdpAudioFormat> ReceiveFormat();
As well as this method on RtpRtcp module:
int32_t RegisterSendPayload(const CodecInst&);
Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}
2018-12-11 12:22:10 +01:00
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// Get payload type and format of the last non-CNG/non-DTMF received payload.
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// If no non-CNG/non-DTMF packet is received std::nullopt is returned.
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//
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std::optional<std::pair<int, SdpAudioFormat>> LastDecoder() const;
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2013-09-12 18:30:26 +00:00
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//
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// Enable NACK and set the maximum size of the NACK list. If NACK is already
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// enabled then the maximum NACK list size is modified accordingly.
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//
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// If the sequence number of last received packet is N, the sequence numbers
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// of NACK list are in the range of [N - `max_nack_list_size`, N).
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//
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// `max_nack_list_size` should be positive (none zero) and less than or
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// equal to `Nack::kNackListSizeLimit`. Otherwise, No change is applied and -1
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// is returned. 0 is returned at success.
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//
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int EnableNack(size_t max_nack_list_size);
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// Disable NACK.
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void DisableNack();
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//
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// Get a list of packets to be retransmitted. `round_trip_time_ms` is an
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// estimate of the round-trip-time (in milliseconds). Missing packets which
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// will be playout in a shorter time than the round-trip-time (with respect
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// to the time this API is called) will not be included in the list.
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//
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// Negative `round_trip_time_ms` results is an error message and empty list
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// is returned.
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//
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std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
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2013-12-13 19:17:43 +00:00
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//
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// Get statistics of calls to GetAudio().
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void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
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2013-09-12 18:30:26 +00:00
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private:
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uint32_t NowInTimestamp(int decoder_sampling_rate) const;
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2024-08-06 16:04:07 +02:00
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const Environment env_;
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2020-07-07 15:53:34 +02:00
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mutable Mutex mutex_;
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CallStatistics call_stats_ RTC_GUARDED_BY(mutex_);
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const std::unique_ptr<NetEq> neteq_; // NetEq is thread-safe; no lock needed.
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2024-09-06 11:53:14 +00:00
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ResamplerHelper resampler_helper_ RTC_GUARDED_BY(mutex_);
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2013-09-12 18:30:26 +00:00
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};
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2013-10-06 04:47:28 +00:00
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} // namespace acm2
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2013-09-12 18:30:26 +00:00
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} // namespace webrtc
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2017-09-15 06:47:31 +02:00
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#endif // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
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