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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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623 if (ret != PayloadSplitter::kOK) { | 624 if (ret != PayloadSplitter::kOK) { |
624 PacketBuffer::DeleteAllPackets(&packet_list); | 625 PacketBuffer::DeleteAllPackets(&packet_list); |
625 switch (ret) { | 626 switch (ret) { |
626 case PayloadSplitter::kUnknownPayloadType: | 627 case PayloadSplitter::kUnknownPayloadType: |
627 return kUnknownRtpPayloadType; | 628 return kUnknownRtpPayloadType; |
628 default: | 629 default: |
629 return kOtherError; | 630 return kOtherError; |
630 } | 631 } |
631 } | 632 } |
632 | 633 |
633 // Split payloads into smaller chunks. This also verifies that all payloads | |
634 // are of a known payload type. | |
635 ret = payload_splitter_->SplitAudio(&packet_list, *decoder_database_); | |
636 if (ret != PayloadSplitter::kOK) { | |
637 PacketBuffer::DeleteAllPackets(&packet_list); | |
638 switch (ret) { | |
639 case PayloadSplitter::kUnknownPayloadType: | |
640 return kUnknownRtpPayloadType; | |
641 case PayloadSplitter::kFrameSplitError: | |
642 return kFrameSplitError; | |
643 default: | |
644 return kOtherError; | |
645 } | |
646 } | |
647 | |
648 // Update bandwidth estimate, if the packet is not comfort noise. | 634 // Update bandwidth estimate, if the packet is not comfort noise. |
649 if (!packet_list.empty() && | 635 if (!packet_list.empty() && |
650 !decoder_database_->IsComfortNoise(main_header.payloadType)) { | 636 !decoder_database_->IsComfortNoise(main_header.payloadType)) { |
651 // The list can be empty here if we got nothing but DTMF payloads. | 637 // The list can be empty here if we got nothing but DTMF payloads. |
652 AudioDecoder* decoder = | 638 AudioDecoder* decoder = |
653 decoder_database_->GetDecoder(main_header.payloadType); | 639 decoder_database_->GetDecoder(main_header.payloadType); |
654 assert(decoder); // Should always get a valid object, since we have | 640 assert(decoder); // Should always get a valid object, since we have |
655 // already checked that the payload types are known. | 641 // already checked that the payload types are known. |
656 decoder->IncomingPacket(packet_list.front()->payload.data(), | 642 decoder->IncomingPacket(packet_list.front()->payload.data(), |
657 packet_list.front()->payload.size(), | 643 packet_list.front()->payload.size(), |
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675 // Carry comfort noise packets along. | 661 // Carry comfort noise packets along. |
676 parsed_packet_list.push_back(packet.release()); | 662 parsed_packet_list.push_back(packet.release()); |
677 } else { | 663 } else { |
678 std::vector<AudioDecoder::ParseResult> results = | 664 std::vector<AudioDecoder::ParseResult> results = |
679 info->GetDecoder()->ParsePayload(std::move(packet->payload), | 665 info->GetDecoder()->ParsePayload(std::move(packet->payload), |
680 packet->header.timestamp, | 666 packet->header.timestamp, |
681 packet->primary); | 667 packet->primary); |
682 const RTPHeader& original_header = packet->header; | 668 const RTPHeader& original_header = packet->header; |
683 for (auto& result : results) { | 669 for (auto& result : results) { |
684 RTC_DCHECK(result.frame); | 670 RTC_DCHECK(result.frame); |
685 // Reuse the packet if possible | 671 // Reuse the packet if possible |
kwiberg-webrtc
2016/09/16 00:48:08
End comment with "." Or, for a less minimalist ton
ossu
2016/09/16 11:46:00
Monoton und minimal.
| |
686 if (!packet) { | 672 if (!packet) { |
687 packet.reset(new Packet); | 673 packet.reset(new Packet); |
688 packet->header = original_header; | 674 packet->header = original_header; |
689 } | 675 } |
kwiberg-webrtc
2016/09/16 00:48:08
Again, should packet->payload (moved from on line
ossu
2016/09/16 11:46:00
I think it's probably best if it's not: once moved
kwiberg-webrtc
2016/09/16 12:07:39
Acknowledged.
| |
690 packet->header.timestamp = result.timestamp; | 676 packet->header.timestamp = result.timestamp; |
691 // TODO(ossu): Move from primary to some sort of priority level. | 677 // TODO(ossu): Move from primary to some sort of priority level. |
692 packet->primary = result.primary; | 678 packet->primary = result.primary; |
693 packet->frame = std::move(result.frame); | 679 packet->frame = std::move(result.frame); |
694 parsed_packet_list.push_back(packet.release()); | 680 parsed_packet_list.push_back(packet.release()); |
695 } | 681 } |
696 } | 682 } |
697 } | 683 } |
698 | 684 |
699 if (nack_enabled_) { | 685 if (nack_enabled_) { |
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2098 } | 2084 } |
2099 } | 2085 } |
2100 | 2086 |
2101 void NetEqImpl::CreateDecisionLogic() { | 2087 void NetEqImpl::CreateDecisionLogic() { |
2102 decision_logic_.reset(DecisionLogic::Create( | 2088 decision_logic_.reset(DecisionLogic::Create( |
2103 fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(), | 2089 fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(), |
2104 *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(), | 2090 *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(), |
2105 tick_timer_.get())); | 2091 tick_timer_.get())); |
2106 } | 2092 } |
2107 } // namespace webrtc | 2093 } // namespace webrtc |
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