Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(84)

Side by Side Diff: webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc

Issue 1949533002: WIP: Move the creation of AudioCodecFactory into PeerConnectionFactory. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 // TODO(hlundin): The functionality in this file should be moved into one or 11 // TODO(hlundin): The functionality in this file should be moved into one or
12 // several classes. 12 // several classes.
13 13
14 #include <assert.h> 14 #include <assert.h>
15 #include <errno.h> 15 #include <errno.h>
16 #include <limits.h> // For ULONG_MAX returned by strtoul. 16 #include <limits.h> // For ULONG_MAX returned by strtoul.
17 #include <stdio.h> 17 #include <stdio.h>
18 #include <stdlib.h> // For strtoul. 18 #include <stdlib.h> // For strtoul.
19 19
20 #include <algorithm> 20 #include <algorithm>
21 #include <iostream> 21 #include <iostream>
22 #include <memory> 22 #include <memory>
23 #include <limits> 23 #include <limits>
24 #include <string> 24 #include <string>
25 25
26 #include "gflags/gflags.h" 26 #include "gflags/gflags.h"
27 #include "webrtc/base/checks.h" 27 #include "webrtc/base/checks.h"
28 #include "webrtc/base/safe_conversions.h" 28 #include "webrtc/base/safe_conversions.h"
29 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
29 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" 30 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
30 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" 31 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
31 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" 32 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
32 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h" 33 #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
33 #include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h" 34 #include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h"
34 #include "webrtc/modules/audio_coding/neteq/tools/packet.h" 35 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
35 #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h" 36 #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
36 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" 37 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
37 #include "webrtc/modules/include/module_common_types.h" 38 #include "webrtc/modules/include/module_common_types.h"
38 #include "webrtc/system_wrappers/include/trace.h" 39 #include "webrtc/system_wrappers/include/trace.h"
(...skipping 442 matching lines...) Expand 10 before | Expand all | Expand 10 after
481 482
482 // Enable tracing. 483 // Enable tracing.
483 webrtc::Trace::CreateTrace(); 484 webrtc::Trace::CreateTrace();
484 webrtc::Trace::SetTraceFile((webrtc::test::OutputPath() + 485 webrtc::Trace::SetTraceFile((webrtc::test::OutputPath() +
485 "neteq_trace.txt").c_str()); 486 "neteq_trace.txt").c_str());
486 webrtc::Trace::set_level_filter(webrtc::kTraceAll); 487 webrtc::Trace::set_level_filter(webrtc::kTraceAll);
487 488
488 // Initialize NetEq instance. 489 // Initialize NetEq instance.
489 NetEq::Config config; 490 NetEq::Config config;
490 config.sample_rate_hz = sample_rate_hz; 491 config.sample_rate_hz = sample_rate_hz;
491 NetEq* neteq = NetEq::Create(config); 492 NetEq* neteq =
493 NetEq::Create(config, webrtc::CreateBuiltinAudioDecoderFactory());
492 RegisterPayloadTypes(neteq); 494 RegisterPayloadTypes(neteq);
493 495
494 496
495 // Set up variables for audio replacement if needed. 497 // Set up variables for audio replacement if needed.
496 std::unique_ptr<webrtc::test::Packet> next_packet; 498 std::unique_ptr<webrtc::test::Packet> next_packet;
497 bool next_packet_available = false; 499 bool next_packet_available = false;
498 size_t input_frame_size_timestamps = 0; 500 size_t input_frame_size_timestamps = 0;
499 std::unique_ptr<int16_t[]> replacement_audio; 501 std::unique_ptr<int16_t[]> replacement_audio;
500 std::unique_ptr<uint8_t[]> payload; 502 std::unique_ptr<uint8_t[]> payload;
501 size_t payload_mem_size_bytes = 0; 503 size_t payload_mem_size_bytes = 0;
(...skipping 131 matching lines...) Expand 10 before | Expand all | Expand 10 after
633 } 635 }
634 } 636 }
635 printf("Simulation done\n"); 637 printf("Simulation done\n");
636 printf("Produced %i ms of audio\n", 638 printf("Produced %i ms of audio\n",
637 static_cast<int>(time_now_ms - start_time_ms)); 639 static_cast<int>(time_now_ms - start_time_ms));
638 640
639 delete neteq; 641 delete neteq;
640 webrtc::Trace::ReturnTrace(); 642 webrtc::Trace::ReturnTrace();
641 return 0; 643 return 0;
642 } 644 }
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698