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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/modules/audio_coding/neteq/neteq_impl.h" | 11 #include "webrtc/modules/audio_coding/neteq/neteq_impl.h" |
12 | 12 |
13 #include <assert.h> | 13 #include <assert.h> |
14 #include <memory.h> // memset | 14 #include <memory.h> // memset |
15 | 15 |
16 #include <algorithm> | 16 #include <algorithm> |
| 17 #include <utility> |
17 #include <vector> | 18 #include <vector> |
18 | 19 |
19 #include "webrtc/base/checks.h" | 20 #include "webrtc/base/checks.h" |
20 #include "webrtc/base/logging.h" | 21 #include "webrtc/base/logging.h" |
21 #include "webrtc/base/safe_conversions.h" | 22 #include "webrtc/base/safe_conversions.h" |
22 #include "webrtc/base/sanitizer.h" | 23 #include "webrtc/base/sanitizer.h" |
23 #include "webrtc/base/trace_event.h" | 24 #include "webrtc/base/trace_event.h" |
24 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" | 25 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" |
25 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" | 26 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
26 #include "webrtc/modules/audio_coding/neteq/accelerate.h" | 27 #include "webrtc/modules/audio_coding/neteq/accelerate.h" |
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275 } | 276 } |
276 return kFail; | 277 return kFail; |
277 } | 278 } |
278 return kOK; | 279 return kOK; |
279 } | 280 } |
280 | 281 |
281 int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) { | 282 int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) { |
282 rtc::CritScope lock(&crit_sect_); | 283 rtc::CritScope lock(&crit_sect_); |
283 int ret = decoder_database_->Remove(rtp_payload_type); | 284 int ret = decoder_database_->Remove(rtp_payload_type); |
284 if (ret == DecoderDatabase::kOK) { | 285 if (ret == DecoderDatabase::kOK) { |
| 286 packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type); |
285 return kOK; | 287 return kOK; |
286 } else if (ret == DecoderDatabase::kDecoderNotFound) { | 288 } else if (ret == DecoderDatabase::kDecoderNotFound) { |
287 error_code_ = kDecoderNotFound; | 289 error_code_ = kDecoderNotFound; |
288 } else { | 290 } else { |
289 error_code_ = kOtherError; | 291 error_code_ = kOtherError; |
290 } | 292 } |
291 return kFail; | 293 return kFail; |
292 } | 294 } |
293 | 295 |
294 bool NetEqImpl::SetMinimumDelay(int delay_ms) { | 296 bool NetEqImpl::SetMinimumDelay(int delay_ms) { |
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323 return kNotImplemented; | 325 return kNotImplemented; |
324 } | 326 } |
325 | 327 |
326 int NetEqImpl::CurrentDelayMs() const { | 328 int NetEqImpl::CurrentDelayMs() const { |
327 rtc::CritScope lock(&crit_sect_); | 329 rtc::CritScope lock(&crit_sect_); |
328 if (fs_hz_ == 0) | 330 if (fs_hz_ == 0) |
329 return 0; | 331 return 0; |
330 // Sum up the samples in the packet buffer with the future length of the sync | 332 // Sum up the samples in the packet buffer with the future length of the sync |
331 // buffer, and divide the sum by the sample rate. | 333 // buffer, and divide the sum by the sample rate. |
332 const size_t delay_samples = | 334 const size_t delay_samples = |
333 packet_buffer_->NumSamplesInBuffer(decoder_database_.get(), | 335 packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) + |
334 decoder_frame_length_) + | |
335 sync_buffer_->FutureLength(); | 336 sync_buffer_->FutureLength(); |
336 // The division below will truncate. | 337 // The division below will truncate. |
337 const int delay_ms = | 338 const int delay_ms = |
338 static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000); | 339 static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000); |
339 return delay_ms; | 340 return delay_ms; |
340 } | 341 } |
341 | 342 |
342 int NetEqImpl::FilteredCurrentDelayMs() const { | 343 int NetEqImpl::FilteredCurrentDelayMs() const { |
343 rtc::CritScope lock(&crit_sect_); | 344 rtc::CritScope lock(&crit_sect_); |
344 // Calculate the filtered packet buffer level in samples. The value from | 345 // Calculate the filtered packet buffer level in samples. The value from |
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369 // TODO(henrik.lundin) Delete. | 370 // TODO(henrik.lundin) Delete. |
370 NetEqPlayoutMode NetEqImpl::PlayoutMode() const { | 371 NetEqPlayoutMode NetEqImpl::PlayoutMode() const { |
371 rtc::CritScope lock(&crit_sect_); | 372 rtc::CritScope lock(&crit_sect_); |
372 return playout_mode_; | 373 return playout_mode_; |
373 } | 374 } |
374 | 375 |
375 int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) { | 376 int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) { |
376 rtc::CritScope lock(&crit_sect_); | 377 rtc::CritScope lock(&crit_sect_); |
377 assert(decoder_database_.get()); | 378 assert(decoder_database_.get()); |
378 const size_t total_samples_in_buffers = | 379 const size_t total_samples_in_buffers = |
379 packet_buffer_->NumSamplesInBuffer(decoder_database_.get(), | 380 packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) + |
380 decoder_frame_length_) + | |
381 sync_buffer_->FutureLength(); | 381 sync_buffer_->FutureLength(); |
382 assert(delay_manager_.get()); | 382 assert(delay_manager_.get()); |
383 assert(decision_logic_.get()); | 383 assert(decision_logic_.get()); |
384 stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers, | 384 stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers, |
385 decoder_frame_length_, *delay_manager_.get(), | 385 decoder_frame_length_, *delay_manager_.get(), |
386 *decision_logic_.get(), stats); | 386 *decision_logic_.get(), stats); |
387 return 0; | 387 return 0; |
388 } | 388 } |
389 | 389 |
390 void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) { | 390 void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) { |
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655 decoder_database_->GetDecoder(main_header.payloadType); | 655 decoder_database_->GetDecoder(main_header.payloadType); |
656 assert(decoder); // Should always get a valid object, since we have | 656 assert(decoder); // Should always get a valid object, since we have |
657 // already checked that the payload types are known. | 657 // already checked that the payload types are known. |
658 decoder->IncomingPacket(packet_list.front()->payload.data(), | 658 decoder->IncomingPacket(packet_list.front()->payload.data(), |
659 packet_list.front()->payload.size(), | 659 packet_list.front()->payload.size(), |
660 packet_list.front()->header.sequenceNumber, | 660 packet_list.front()->header.sequenceNumber, |
661 packet_list.front()->header.timestamp, | 661 packet_list.front()->header.timestamp, |
662 receive_timestamp); | 662 receive_timestamp); |
663 } | 663 } |
664 | 664 |
| 665 PacketList parsed_packet_list; |
| 666 while (!packet_list.empty()) { |
| 667 std::unique_ptr<Packet> packet(packet_list.front()); |
| 668 packet_list.pop_front(); |
| 669 const DecoderDatabase::DecoderInfo* info = |
| 670 decoder_database_->GetDecoderInfo(packet->header.payloadType); |
| 671 if (!info) { |
| 672 LOG(LS_WARNING) << "SplitAudio unknown payload type"; |
| 673 return kUnknownRtpPayloadType; |
| 674 } |
| 675 |
| 676 if (info->IsComfortNoise()) { |
| 677 // Carry comfort noise packets along. |
| 678 parsed_packet_list.push_back(packet.release()); |
| 679 } else { |
| 680 std::vector<AudioDecoder::ParseResult> results = |
| 681 info->GetDecoder()->ParsePayload(std::move(packet->payload), |
| 682 packet->header.timestamp, |
| 683 packet->primary); |
| 684 const RTPHeader& original_header = packet->header; |
| 685 for (auto& result : results) { |
| 686 RTC_DCHECK(result.frame); |
| 687 // Reuse the packet if possible |
| 688 if (!packet) { |
| 689 packet.reset(new Packet); |
| 690 packet->header = original_header; |
| 691 } |
| 692 packet->header.timestamp = result.timestamp; |
| 693 // TODO(ossu): Move from primary to some sort of priority level. |
| 694 packet->primary = result.primary; |
| 695 packet->frame = std::move(result.frame); |
| 696 parsed_packet_list.push_back(packet.release()); |
| 697 } |
| 698 } |
| 699 } |
| 700 |
665 if (nack_enabled_) { | 701 if (nack_enabled_) { |
666 RTC_DCHECK(nack_); | 702 RTC_DCHECK(nack_); |
667 if (update_sample_rate_and_channels) { | 703 if (update_sample_rate_and_channels) { |
668 nack_->Reset(); | 704 nack_->Reset(); |
669 } | 705 } |
670 nack_->UpdateLastReceivedPacket(packet_list.front()->header.sequenceNumber, | 706 nack_->UpdateLastReceivedPacket( |
671 packet_list.front()->header.timestamp); | 707 parsed_packet_list.front()->header.sequenceNumber, |
| 708 parsed_packet_list.front()->header.timestamp); |
672 } | 709 } |
673 | 710 |
674 // Insert packets in buffer. | 711 // Insert packets in buffer. |
675 const size_t buffer_length_before_insert = | 712 const size_t buffer_length_before_insert = |
676 packet_buffer_->NumPacketsInBuffer(); | 713 packet_buffer_->NumPacketsInBuffer(); |
677 ret = packet_buffer_->InsertPacketList( | 714 ret = packet_buffer_->InsertPacketList( |
678 &packet_list, | 715 &parsed_packet_list, *decoder_database_, ¤t_rtp_payload_type_, |
679 *decoder_database_, | |
680 ¤t_rtp_payload_type_, | |
681 ¤t_cng_rtp_payload_type_); | 716 ¤t_cng_rtp_payload_type_); |
682 if (ret == PacketBuffer::kFlushed) { | 717 if (ret == PacketBuffer::kFlushed) { |
683 // Reset DSP timestamp etc. if packet buffer flushed. | 718 // Reset DSP timestamp etc. if packet buffer flushed. |
684 new_codec_ = true; | 719 new_codec_ = true; |
685 update_sample_rate_and_channels = true; | 720 update_sample_rate_and_channels = true; |
686 } else if (ret != PacketBuffer::kOK) { | 721 } else if (ret != PacketBuffer::kOK) { |
687 PacketBuffer::DeleteAllPackets(&packet_list); | 722 PacketBuffer::DeleteAllPackets(&parsed_packet_list); |
688 return kOtherError; | 723 return kOtherError; |
689 } | 724 } |
690 | 725 |
691 if (first_packet_) { | 726 if (first_packet_) { |
692 first_packet_ = false; | 727 first_packet_ = false; |
693 // Update the codec on the next GetAudio call. | 728 // Update the codec on the next GetAudio call. |
694 new_codec_ = true; | 729 new_codec_ = true; |
695 } | 730 } |
696 | 731 |
697 if (current_rtp_payload_type_) { | 732 if (current_rtp_payload_type_) { |
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1414 !decoder_database_->IsComfortNoise(packet->header.payloadType)) { | 1449 !decoder_database_->IsComfortNoise(packet->header.payloadType)) { |
1415 assert(decoder); // At this point, we must have a decoder object. | 1450 assert(decoder); // At this point, we must have a decoder object. |
1416 // The number of channels in the |sync_buffer_| should be the same as the | 1451 // The number of channels in the |sync_buffer_| should be the same as the |
1417 // number decoder channels. | 1452 // number decoder channels. |
1418 assert(sync_buffer_->Channels() == decoder->Channels()); | 1453 assert(sync_buffer_->Channels() == decoder->Channels()); |
1419 assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->Channels()); | 1454 assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->Channels()); |
1420 assert(operation == kNormal || operation == kAccelerate || | 1455 assert(operation == kNormal || operation == kAccelerate || |
1421 operation == kFastAccelerate || operation == kMerge || | 1456 operation == kFastAccelerate || operation == kMerge || |
1422 operation == kPreemptiveExpand); | 1457 operation == kPreemptiveExpand); |
1423 packet_list->pop_front(); | 1458 packet_list->pop_front(); |
1424 const size_t payload_length = packet->payload.size(); | 1459 auto opt_result = packet->frame->Decode( |
1425 int decode_length; | 1460 rtc::ArrayView<int16_t>(&decoded_buffer_[*decoded_length], |
1426 if (!packet->primary) { | 1461 decoded_buffer_length_ - *decoded_length)); |
1427 // This is a redundant payload; call the special decoder method. | |
1428 decode_length = decoder->DecodeRedundant( | |
1429 packet->payload.data(), packet->payload.size(), fs_hz_, | |
1430 (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t), | |
1431 &decoded_buffer_[*decoded_length], speech_type); | |
1432 } else { | |
1433 decode_length = decoder->Decode( | |
1434 packet->payload.data(), packet->payload.size(), fs_hz_, | |
1435 (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t), | |
1436 &decoded_buffer_[*decoded_length], speech_type); | |
1437 } | |
1438 | |
1439 delete packet; | 1462 delete packet; |
1440 packet = NULL; | 1463 packet = NULL; |
1441 if (decode_length > 0) { | 1464 if (opt_result) { |
1442 *decoded_length += decode_length; | 1465 const auto& result = *opt_result; |
1443 // Update |decoder_frame_length_| with number of samples per channel. | 1466 *speech_type = result.speech_type; |
1444 decoder_frame_length_ = | 1467 if (result.num_decoded_samples > 0) { |
1445 static_cast<size_t>(decode_length) / decoder->Channels(); | 1468 *decoded_length += rtc::checked_cast<int>(result.num_decoded_samples); |
1446 } else if (decode_length < 0) { | 1469 // Update |decoder_frame_length_| with number of samples per channel. |
| 1470 decoder_frame_length_ = |
| 1471 result.num_decoded_samples / decoder->Channels(); |
| 1472 } |
| 1473 } else { |
1447 // Error. | 1474 // Error. |
1448 LOG(LS_WARNING) << "Decode " << decode_length << " " << payload_length; | 1475 // TODO(ossu): What to put here? |
| 1476 LOG(LS_WARNING) << "Decode error"; |
1449 *decoded_length = -1; | 1477 *decoded_length = -1; |
1450 PacketBuffer::DeleteAllPackets(packet_list); | 1478 PacketBuffer::DeleteAllPackets(packet_list); |
1451 break; | 1479 break; |
1452 } | 1480 } |
1453 if (*decoded_length > static_cast<int>(decoded_buffer_length_)) { | 1481 if (*decoded_length > rtc::checked_cast<int>(decoded_buffer_length_)) { |
1454 // Guard against overflow. | 1482 // Guard against overflow. |
1455 LOG(LS_WARNING) << "Decoded too much."; | 1483 LOG(LS_WARNING) << "Decoded too much."; |
1456 PacketBuffer::DeleteAllPackets(packet_list); | 1484 PacketBuffer::DeleteAllPackets(packet_list); |
1457 return kDecodedTooMuch; | 1485 return kDecodedTooMuch; |
1458 } | 1486 } |
1459 if (!packet_list->empty()) { | 1487 if (!packet_list->empty()) { |
1460 packet = packet_list->front(); | 1488 packet = packet_list->front(); |
1461 } else { | 1489 } else { |
1462 packet = NULL; | 1490 packet = NULL; |
1463 } | 1491 } |
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1885 uint16_t prev_sequence_number = 0; | 1913 uint16_t prev_sequence_number = 0; |
1886 bool next_packet_available = false; | 1914 bool next_packet_available = false; |
1887 | 1915 |
1888 const RTPHeader* header = packet_buffer_->NextRtpHeader(); | 1916 const RTPHeader* header = packet_buffer_->NextRtpHeader(); |
1889 assert(header); | 1917 assert(header); |
1890 if (!header) { | 1918 if (!header) { |
1891 LOG(LS_ERROR) << "Packet buffer unexpectedly empty."; | 1919 LOG(LS_ERROR) << "Packet buffer unexpectedly empty."; |
1892 return -1; | 1920 return -1; |
1893 } | 1921 } |
1894 uint32_t first_timestamp = header->timestamp; | 1922 uint32_t first_timestamp = header->timestamp; |
1895 int extracted_samples = 0; | 1923 size_t extracted_samples = 0; |
1896 | 1924 |
1897 // Packet extraction loop. | 1925 // Packet extraction loop. |
1898 do { | 1926 do { |
1899 timestamp_ = header->timestamp; | 1927 timestamp_ = header->timestamp; |
1900 size_t discard_count = 0; | 1928 size_t discard_count = 0; |
1901 Packet* packet = packet_buffer_->GetNextPacket(&discard_count); | 1929 Packet* packet = packet_buffer_->GetNextPacket(&discard_count); |
1902 // |header| may be invalid after the |packet_buffer_| operation. | 1930 // |header| may be invalid after the |packet_buffer_| operation. |
1903 header = NULL; | 1931 header = NULL; |
1904 if (!packet) { | 1932 if (!packet) { |
1905 LOG(LS_ERROR) << "Should always be able to extract a packet here"; | 1933 LOG(LS_ERROR) << "Should always be able to extract a packet here"; |
1906 assert(false); // Should always be able to extract a packet here. | 1934 assert(false); // Should always be able to extract a packet here. |
1907 return -1; | 1935 return -1; |
1908 } | 1936 } |
1909 stats_.PacketsDiscarded(discard_count); | 1937 stats_.PacketsDiscarded(discard_count); |
1910 stats_.StoreWaitingTime(packet->waiting_time->ElapsedMs()); | 1938 stats_.StoreWaitingTime(packet->waiting_time->ElapsedMs()); |
1911 assert(!packet->payload.empty()); | 1939 RTC_DCHECK(!packet->empty()); |
1912 packet_list->push_back(packet); // Store packet in list. | 1940 packet_list->push_back(packet); // Store packet in list. |
1913 | 1941 |
1914 if (first_packet) { | 1942 if (first_packet) { |
1915 first_packet = false; | 1943 first_packet = false; |
1916 if (nack_enabled_) { | 1944 if (nack_enabled_) { |
1917 RTC_DCHECK(nack_); | 1945 RTC_DCHECK(nack_); |
1918 // TODO(henrik.lundin): Should we update this for all decoded packets? | 1946 // TODO(henrik.lundin): Should we update this for all decoded packets? |
1919 nack_->UpdateLastDecodedPacket(packet->header.sequenceNumber, | 1947 nack_->UpdateLastDecodedPacket(packet->header.sequenceNumber, |
1920 packet->header.timestamp); | 1948 packet->header.timestamp); |
1921 } | 1949 } |
1922 prev_sequence_number = packet->header.sequenceNumber; | 1950 prev_sequence_number = packet->header.sequenceNumber; |
1923 prev_timestamp = packet->header.timestamp; | 1951 prev_timestamp = packet->header.timestamp; |
1924 prev_payload_type = packet->header.payloadType; | 1952 prev_payload_type = packet->header.payloadType; |
1925 } | 1953 } |
1926 | 1954 |
1927 // Store number of extracted samples. | 1955 // Store number of extracted samples. |
1928 int packet_duration = 0; | 1956 size_t packet_duration = 0; |
1929 AudioDecoder* decoder = decoder_database_->GetDecoder( | 1957 if (packet->frame) { |
1930 packet->header.payloadType); | 1958 packet_duration = packet->frame->Duration(); |
1931 if (decoder) { | 1959 // TODO(ossu): Is this the correct way to track samples decoded from a |
1932 if (packet->primary) { | 1960 // redundant packet? |
1933 packet_duration = decoder->PacketDuration(packet->payload.data(), | 1961 if (packet_duration > 0 && !packet->primary) { |
1934 packet->payload.size()); | 1962 stats_.SecondaryDecodedSamples(rtc::checked_cast<int>(packet_duration)); |
1935 } else { | |
1936 packet_duration = decoder->PacketDurationRedundant( | |
1937 packet->payload.data(), packet->payload.size()); | |
1938 stats_.SecondaryDecodedSamples(packet_duration); | |
1939 } | 1963 } |
1940 } else if (!decoder_database_->IsComfortNoise(packet->header.payloadType)) { | 1964 } else if (!decoder_database_->IsComfortNoise(packet->header.payloadType)) { |
1941 LOG(LS_WARNING) << "Unknown payload type " | 1965 LOG(LS_WARNING) << "Unknown payload type " |
1942 << static_cast<int>(packet->header.payloadType); | 1966 << static_cast<int>(packet->header.payloadType); |
1943 assert(false); | 1967 RTC_NOTREACHED(); |
1944 } | 1968 } |
1945 if (packet_duration <= 0) { | 1969 |
| 1970 if (packet_duration == 0) { |
1946 // Decoder did not return a packet duration. Assume that the packet | 1971 // Decoder did not return a packet duration. Assume that the packet |
1947 // contains the same number of samples as the previous one. | 1972 // contains the same number of samples as the previous one. |
1948 packet_duration = rtc::checked_cast<int>(decoder_frame_length_); | 1973 packet_duration = decoder_frame_length_; |
1949 } | 1974 } |
1950 extracted_samples = packet->header.timestamp - first_timestamp + | 1975 extracted_samples = packet->header.timestamp - first_timestamp + |
1951 packet_duration; | 1976 packet_duration; |
1952 | 1977 |
1953 // Check what packet is available next. | 1978 // Check what packet is available next. |
1954 header = packet_buffer_->NextRtpHeader(); | 1979 header = packet_buffer_->NextRtpHeader(); |
1955 next_packet_available = false; | 1980 next_packet_available = false; |
1956 if (header && prev_payload_type == header->payloadType) { | 1981 if (header && prev_payload_type == header->payloadType) { |
1957 int16_t seq_no_diff = header->sequenceNumber - prev_sequence_number; | 1982 int16_t seq_no_diff = header->sequenceNumber - prev_sequence_number; |
1958 size_t ts_diff = header->timestamp - prev_timestamp; | 1983 size_t ts_diff = header->timestamp - prev_timestamp; |
1959 if (seq_no_diff == 1 || | 1984 if (seq_no_diff == 1 || |
1960 (seq_no_diff == 0 && ts_diff == decoder_frame_length_)) { | 1985 (seq_no_diff == 0 && ts_diff == decoder_frame_length_)) { |
1961 // The next sequence number is available, or the next part of a packet | 1986 // The next sequence number is available, or the next part of a packet |
1962 // that was split into pieces upon insertion. | 1987 // that was split into pieces upon insertion. |
1963 next_packet_available = true; | 1988 next_packet_available = true; |
1964 } | 1989 } |
1965 prev_sequence_number = header->sequenceNumber; | 1990 prev_sequence_number = header->sequenceNumber; |
1966 } | 1991 } |
1967 } while (extracted_samples < rtc::checked_cast<int>(required_samples) && | 1992 } while (extracted_samples < required_samples && next_packet_available); |
1968 next_packet_available); | |
1969 | 1993 |
1970 if (extracted_samples > 0) { | 1994 if (extracted_samples > 0) { |
1971 // Delete old packets only when we are going to decode something. Otherwise, | 1995 // Delete old packets only when we are going to decode something. Otherwise, |
1972 // we could end up in the situation where we never decode anything, since | 1996 // we could end up in the situation where we never decode anything, since |
1973 // all incoming packets are considered too old but the buffer will also | 1997 // all incoming packets are considered too old but the buffer will also |
1974 // never be flooded and flushed. | 1998 // never be flooded and flushed. |
1975 packet_buffer_->DiscardAllOldPackets(timestamp_); | 1999 packet_buffer_->DiscardAllOldPackets(timestamp_); |
1976 } | 2000 } |
1977 | 2001 |
1978 return extracted_samples; | 2002 return rtc::checked_cast<int>(extracted_samples); |
1979 } | 2003 } |
1980 | 2004 |
1981 void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) { | 2005 void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) { |
1982 // Delete objects and create new ones. | 2006 // Delete objects and create new ones. |
1983 expand_.reset(expand_factory_->Create(background_noise_.get(), | 2007 expand_.reset(expand_factory_->Create(background_noise_.get(), |
1984 sync_buffer_.get(), &random_vector_, | 2008 sync_buffer_.get(), &random_vector_, |
1985 &stats_, fs_hz, channels)); | 2009 &stats_, fs_hz, channels)); |
1986 merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get())); | 2010 merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get())); |
1987 } | 2011 } |
1988 | 2012 |
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2075 } | 2099 } |
2076 } | 2100 } |
2077 | 2101 |
2078 void NetEqImpl::CreateDecisionLogic() { | 2102 void NetEqImpl::CreateDecisionLogic() { |
2079 decision_logic_.reset(DecisionLogic::Create( | 2103 decision_logic_.reset(DecisionLogic::Create( |
2080 fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(), | 2104 fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(), |
2081 *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(), | 2105 *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(), |
2082 tick_timer_.get())); | 2106 tick_timer_.get())); |
2083 } | 2107 } |
2084 } // namespace webrtc | 2108 } // namespace webrtc |
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