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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#include "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 <stdlib.h>
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#include <string.h>
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2019-07-05 19:08:33 +02:00
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2018-10-23 12:03:01 +02:00
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#include <cstdint>
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2013-09-12 18:30:26 +00:00
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#include <vector>
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2018-10-22 09:48:08 +02:00
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#include "absl/strings/match.h"
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2018-10-23 12:03:01 +02:00
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#include "api/audio/audio_frame.h"
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2017-09-15 06:47:31 +02:00
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#include "api/audio_codecs/audio_decoder.h"
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2019-10-31 14:38:11 +01:00
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#include "api/neteq/neteq.h"
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2024-05-27 11:22:27 +02:00
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#include "api/units/timestamp.h"
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2017-09-15 06:47:31 +02:00
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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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2019-11-26 12:29:05 +01:00
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#include "modules/audio_coding/neteq/default_neteq_factory.h"
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2017-09-15 06:47:31 +02:00
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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2017-11-22 10:42:26 +01:00
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#include "rtc_base/numerics/safe_conversions.h"
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2018-04-03 13:40:05 +02:00
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#include "rtc_base/strings/audio_format_to_string.h"
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2017-09-15 06:47:31 +02:00
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#include "system_wrappers/include/clock.h"
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2013-09-12 18:30:26 +00:00
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namespace webrtc {
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2013-10-06 04:47:28 +00:00
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namespace acm2 {
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2019-10-31 14:38:11 +01:00
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namespace {
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std::unique_ptr<NetEq> CreateNetEq(
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2019-11-01 11:47:51 +01:00
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NetEqFactory* neteq_factory,
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2019-10-31 14:38:11 +01:00
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const NetEq::Config& config,
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2024-08-06 17:50:04 +02:00
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const Environment& env,
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scoped_refptr<AudioDecoderFactory> decoder_factory) {
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2019-11-01 11:47:51 +01:00
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if (neteq_factory) {
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2024-08-06 17:50:04 +02:00
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return neteq_factory->Create(env, config, std::move(decoder_factory));
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2019-11-01 11:47:51 +01:00
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}
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2024-08-06 17:50:04 +02:00
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return DefaultNetEqFactory().Create(env, config, std::move(decoder_factory));
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2019-10-31 14:38:11 +01:00
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}
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} // namespace
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2023-01-31 08:40:56 +00:00
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AcmReceiver::Config::Config(
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
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2024-08-05 12:40:46 +02:00
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: decoder_factory(decoder_factory) {}
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2023-01-31 08:40:56 +00:00
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AcmReceiver::Config::Config(const Config&) = default;
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AcmReceiver::Config::~Config() = default;
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2024-08-05 12:40:46 +02:00
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AcmReceiver::AcmReceiver(const Environment& env, Config config)
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2024-08-06 16:04:07 +02:00
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: env_(env),
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2024-08-05 12:40:46 +02:00
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neteq_(CreateNetEq(config.neteq_factory,
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config.neteq_config,
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2024-08-06 17:50:04 +02:00
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env_,
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2024-09-06 11:53:14 +00:00
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std::move(config.decoder_factory))) {}
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2013-09-12 18:30:26 +00:00
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2017-06-14 14:13:02 +02:00
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AcmReceiver::~AcmReceiver() = default;
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2013-09-12 18:30:26 +00:00
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int AcmReceiver::SetMinimumDelay(int delay_ms) {
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if (neteq_->SetMinimumDelay(delay_ms))
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return 0;
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2021-11-24 10:01:32 +00:00
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RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
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2013-09-12 18:30:26 +00:00
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return -1;
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}
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int AcmReceiver::SetMaximumDelay(int delay_ms) {
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if (neteq_->SetMaximumDelay(delay_ms))
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return 0;
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2021-11-24 10:01:32 +00:00
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RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
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2013-09-12 18:30:26 +00:00
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return -1;
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}
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2019-02-05 13:49:26 +01:00
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bool AcmReceiver::SetBaseMinimumDelayMs(int delay_ms) {
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return neteq_->SetBaseMinimumDelayMs(delay_ms);
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}
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int AcmReceiver::GetBaseMinimumDelayMs() const {
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return neteq_->GetBaseMinimumDelayMs();
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}
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2024-08-29 13:00:40 +00:00
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std::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
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std::optional<NetEq::DecoderFormat> decoder =
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2024-08-28 14:35:34 +00:00
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neteq_->GetCurrentDecoderFormat();
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if (!decoder) {
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2024-08-29 13:00:40 +00:00
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return std::nullopt;
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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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}
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2024-08-28 14:35:34 +00:00
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return decoder->sample_rate_hz;
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2015-11-23 08:19:52 -08:00
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}
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2015-11-23 06:49:25 -08:00
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int AcmReceiver::last_output_sample_rate_hz() const {
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return neteq_->last_output_sample_rate_hz();
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2013-09-12 18:30:26 +00:00
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}
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2019-02-15 15:21:47 +01:00
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int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
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2024-05-27 11:22:27 +02:00
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rtc::ArrayView<const uint8_t> incoming_payload,
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Timestamp receive_time) {
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Handle padded audio packets correctly
RTP packets can be padded with extra data at the end of the payload. The usable
payload length of the packet should then be reduced with the padding length,
since the padding must be discarded. This was not the case; instead, the entire
payload, including padding data, was forwarded to the audio channel and in the
end to the decoder.
A special case of padding is packets which are empty except for the padding.
That is, they carry no usable payload. These packets are sometimes used for
probing the network and were discarded in
RTPReceiverAudio::ParseAudioCodecSpecific. The result is that NetEq never sees
those empty packets, just the holes in the sequence number series; this can
throw off the target buffer calculations.
With this change, the empty (after removing the padding) packets are let through,
all the way down to NetEq, to a new method called NetEq::InsertEmptyPacket. This
method notifies the DelayManager that an empty packet was received.
BUG=webrtc:7610, webrtc:7625
Review-Url: https://codereview.webrtc.org/2870043003
Cr-Commit-Position: refs/heads/master@{#18083}
2017-05-10 07:38:01 -07:00
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if (incoming_payload.empty()) {
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2019-02-15 15:21:47 +01:00
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neteq_->InsertEmptyPacket(rtp_header);
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Handle padded audio packets correctly
RTP packets can be padded with extra data at the end of the payload. The usable
payload length of the packet should then be reduced with the padding length,
since the padding must be discarded. This was not the case; instead, the entire
payload, including padding data, was forwarded to the audio channel and in the
end to the decoder.
A special case of padding is packets which are empty except for the padding.
That is, they carry no usable payload. These packets are sometimes used for
probing the network and were discarded in
RTPReceiverAudio::ParseAudioCodecSpecific. The result is that NetEq never sees
those empty packets, just the holes in the sequence number series; this can
throw off the target buffer calculations.
With this change, the empty (after removing the padding) packets are let through,
all the way down to NetEq, to a new method called NetEq::InsertEmptyPacket. This
method notifies the DelayManager that an empty packet was received.
BUG=webrtc:7610, webrtc:7625
Review-Url: https://codereview.webrtc.org/2870043003
Cr-Commit-Position: refs/heads/master@{#18083}
2017-05-10 07:38:01 -07:00
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return 0;
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}
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2024-05-27 11:22:27 +02:00
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if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_time) < 0) {
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2021-11-24 10:01:32 +00:00
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RTC_LOG(LS_ERROR) << "AcmReceiver::InsertPacket "
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<< static_cast<int>(rtp_header.payloadType)
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<< " Failed to insert packet";
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2014-06-24 13:11:22 +00:00
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return -1;
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2013-09-12 18:30:26 +00:00
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}
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return 0;
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}
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2016-05-13 03:45:24 -07:00
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int AcmReceiver::GetAudio(int desired_freq_hz,
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AudioFrame* audio_frame,
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bool* muted) {
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2021-06-09 19:30:41 +02:00
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int current_sample_rate_hz = 0;
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if (neteq_->GetAudio(audio_frame, muted, ¤t_sample_rate_hz) !=
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NetEq::kOK) {
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2021-11-24 10:01:32 +00:00
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RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
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2014-06-24 13:11:22 +00:00
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return -1;
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2013-09-12 18:30:26 +00:00
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}
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2024-09-06 11:53:14 +00:00
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RTC_DCHECK_EQ(audio_frame->sample_rate_hz_, current_sample_rate_hz);
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2013-09-12 18:30:26 +00:00
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2021-06-09 19:30:41 +02:00
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// Accessing members, take the lock.
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MutexLock lock(&mutex_);
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2024-09-06 11:53:14 +00:00
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if (!resampler_helper_.MaybeResample(desired_freq_hz, audio_frame)) {
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return -1;
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2013-09-12 18:30:26 +00:00
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}
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2024-05-06 16:46:48 +02:00
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call_stats_.DecodedByNetEq(audio_frame->speech_type_, audio_frame->muted());
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2013-09-12 18:30:26 +00:00
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return 0;
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}
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2017-03-27 07:15:49 -07:00
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void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
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neteq_->SetCodecs(codecs);
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}
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2013-09-12 18:30:26 +00:00
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void AcmReceiver::FlushBuffers() {
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neteq_->FlushBuffers();
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}
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2024-08-29 13:00:40 +00:00
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std::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
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2016-04-06 01:39:22 -07:00
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return neteq_->GetPlayoutTimestamp();
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2013-09-12 18:30:26 +00:00
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}
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2016-08-22 15:39:53 -07:00
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int AcmReceiver::FilteredCurrentDelayMs() const {
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return neteq_->FilteredCurrentDelayMs();
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}
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2017-11-29 09:14:04 +01:00
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int AcmReceiver::TargetDelayMs() const {
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return neteq_->TargetDelayMs();
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}
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2024-08-29 13:00:40 +00:00
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std::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder() const {
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std::optional<NetEq::DecoderFormat> decoder =
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2024-08-28 14:35:34 +00:00
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neteq_->GetCurrentDecoderFormat();
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if (!decoder) {
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2024-08-29 13:00:40 +00:00
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return std::nullopt;
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2013-09-12 18:30:26 +00:00
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}
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2024-08-28 14:35:34 +00:00
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return std::make_pair(decoder->payload_type, decoder->sdp_format);
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2016-10-12 11:04:10 -07:00
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}
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2020-09-14 10:47:50 +02:00
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void AcmReceiver::GetNetworkStatistics(
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NetworkStatistics* acm_stat,
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bool get_and_clear_legacy_stats /* = true */) const {
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2013-09-12 18:30:26 +00:00
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NetEqNetworkStatistics neteq_stat;
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2020-09-14 10:47:50 +02:00
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if (get_and_clear_legacy_stats) {
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// NetEq function always returns zero, so we don't check the return value.
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neteq_->NetworkStatistics(&neteq_stat);
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acm_stat->currentExpandRate = neteq_stat.expand_rate;
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acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
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acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
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acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
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acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
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acm_stat->currentSecondaryDiscardedRate =
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neteq_stat.secondary_discarded_rate;
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acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
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acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
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} else {
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neteq_stat = neteq_->CurrentNetworkStatistics();
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acm_stat->currentExpandRate = 0;
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acm_stat->currentSpeechExpandRate = 0;
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acm_stat->currentPreemptiveRate = 0;
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acm_stat->currentAccelerateRate = 0;
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acm_stat->currentSecondaryDecodedRate = 0;
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acm_stat->currentSecondaryDiscardedRate = 0;
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acm_stat->meanWaitingTimeMs = -1;
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acm_stat->maxWaitingTimeMs = 1;
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}
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2013-09-12 18:30:26 +00:00
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acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
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acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
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2013-09-19 00:12:23 +00:00
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acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
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2017-08-24 17:15:13 -07:00
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NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
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acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
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acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
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2019-04-30 09:45:21 +02:00
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acm_stat->silentConcealedSamples =
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neteq_lifetime_stat.silent_concealed_samples;
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2017-09-18 09:28:20 +02:00
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acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
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2017-10-02 12:00:34 +02:00
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acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
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2020-03-11 11:18:54 +01:00
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acm_stat->jitterBufferTargetDelayMs =
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neteq_lifetime_stat.jitter_buffer_target_delay_ms;
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2022-07-19 16:33:10 +02:00
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acm_stat->jitterBufferMinimumDelayMs =
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neteq_lifetime_stat.jitter_buffer_minimum_delay_ms;
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2019-01-15 15:46:29 +01:00
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acm_stat->jitterBufferEmittedCount =
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neteq_lifetime_stat.jitter_buffer_emitted_count;
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2018-11-27 12:52:16 +01:00
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acm_stat->delayedPacketOutageSamples =
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neteq_lifetime_stat.delayed_packet_outage_samples;
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2019-03-06 09:18:40 +01:00
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acm_stat->relativePacketArrivalDelayMs =
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neteq_lifetime_stat.relative_packet_arrival_delay_ms;
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2019-04-29 17:00:46 +02:00
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acm_stat->interruptionCount = neteq_lifetime_stat.interruption_count;
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acm_stat->totalInterruptionDurationMs =
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neteq_lifetime_stat.total_interruption_duration_ms;
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2019-04-30 09:45:21 +02:00
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acm_stat->insertedSamplesForDeceleration =
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neteq_lifetime_stat.inserted_samples_for_deceleration;
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acm_stat->removedSamplesForAcceleration =
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neteq_lifetime_stat.removed_samples_for_acceleration;
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acm_stat->fecPacketsReceived = neteq_lifetime_stat.fec_packets_received;
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acm_stat->fecPacketsDiscarded = neteq_lifetime_stat.fec_packets_discarded;
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2024-06-04 10:05:31 +02:00
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acm_stat->totalProcessingDelayUs =
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neteq_lifetime_stat.total_processing_delay_us;
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2022-05-25 22:00:14 +02:00
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acm_stat->packetsDiscarded = neteq_lifetime_stat.packets_discarded;
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2018-11-22 17:21:10 +01:00
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NetEqOperationsAndState neteq_operations_and_state =
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neteq_->GetOperationsAndState();
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acm_stat->packetBufferFlushes =
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neteq_operations_and_state.packet_buffer_flushes;
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2013-09-12 18:30:26 +00:00
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}
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int AcmReceiver::EnableNack(size_t max_nack_list_size) {
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2015-10-29 05:36:24 -07:00
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neteq_->EnableNack(max_nack_list_size);
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return 0;
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2013-09-12 18:30:26 +00:00
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}
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void AcmReceiver::DisableNack() {
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2015-10-29 05:36:24 -07:00
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neteq_->DisableNack();
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2013-09-12 18:30:26 +00:00
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}
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std::vector<uint16_t> AcmReceiver::GetNackList(
|
2015-01-12 21:51:21 +00:00
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|
|
int64_t round_trip_time_ms) const {
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2015-10-29 05:36:24 -07:00
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return neteq_->GetNackList(round_trip_time_ms);
|
2013-09-12 18:30:26 +00:00
|
|
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}
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void AcmReceiver::ResetInitialDelay() {
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|
|
|
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neteq_->SetMinimumDelay(0);
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|
|
|
|
// TODO(turajs): Should NetEq Buffer be flushed?
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|
|
|
|
}
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|
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|
|
uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
|
|
|
|
|
// Down-cast the time to (32-6)-bit since we only care about
|
|
|
|
|
// the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
|
|
|
|
|
// We masked 6 most significant bits of 32-bit so there is no overflow in
|
|
|
|
|
// the conversion from milliseconds to timestamp.
|
|
|
|
|
const uint32_t now_in_ms =
|
2024-08-06 16:04:07 +02:00
|
|
|
static_cast<uint32_t>(env_.clock().TimeInMilliseconds() & 0x03ffffff);
|
2013-09-12 18:30:26 +00:00
|
|
|
return static_cast<uint32_t>((decoder_sampling_rate / 1000) * now_in_ms);
|
|
|
|
|
}
|
|
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|
|
2013-12-13 19:17:43 +00:00
|
|
|
void AcmReceiver::GetDecodingCallStatistics(
|
|
|
|
|
AudioDecodingCallStats* stats) const {
|
2020-07-07 15:53:34 +02:00
|
|
|
MutexLock lock(&mutex_);
|
2013-12-13 19:17:43 +00:00
|
|
|
*stats = call_stats_.GetDecodingStatistics();
|
|
|
|
|
}
|
|
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|
2013-10-06 04:47:28 +00:00
|
|
|
} // namespace acm2
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|
2013-09-12 18:30:26 +00:00
|
|
|
} // namespace webrtc
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