Henrik Lundin 1391bd472c Replacing the legacy tool RTPencode with a new rtp_encode ...

This new tool provides the same functionality as the legacy tool, but it
is implemented using AudioCodingModule and AudioEncoder APIs instead of
the naked codecs.

Bug: webrtc:2692
Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc
Reviewed-on: https://webrtc-review.googlesource.com/24861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20857}

2017-11-24 09:05:48 +00:00

..

audio_checksum.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

audio_loop.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_loop.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

audio_sink.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

audio_sink.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

constant_pcm_packet_source.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

constant_pcm_packet_source.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

DEPS

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

encode_neteq_input.cc

Move some numeric utility code from rtc_base/ to rtc_base/numerics/

2017-11-22 11:21:47 +00:00

encode_neteq_input.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

fake_decode_from_file.cc

Move some numeric utility code from rtc_base/ to rtc_base/numerics/

2017-11-22 11:21:47 +00:00

fake_decode_from_file.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

input_audio_file_unittest.cc

Move some numeric utility code from rtc_base/ to rtc_base/numerics/

2017-11-22 11:21:47 +00:00

input_audio_file.cc

Replacing the legacy tool RTPencode with a new rtp_encode

2017-11-24 09:05:48 +00:00

input_audio_file.h

Replacing the legacy tool RTPencode with a new rtp_encode

2017-11-24 09:05:48 +00:00

neteq_delay_analyzer.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

neteq_delay_analyzer.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

neteq_external_decoder_test.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

neteq_external_decoder_test.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

neteq_input.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

neteq_input.h

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

neteq_packet_source_input.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

neteq_packet_source_input.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

neteq_performance_test.cc

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

neteq_performance_test.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

neteq_quality_test.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

neteq_quality_test.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

neteq_replacement_input.cc

Optional: Use nullopt and implicit construction in /modules/audio_coding

2017-11-17 11:58:37 +00:00

neteq_replacement_input.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

neteq_rtpplay.cc

neteq_rtpplay: Add buffer size (target and current) to print-out

2017-11-20 08:07:30 +00:00

neteq_test.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

neteq_test.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

output_audio_file.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

output_wav_file.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

packet_source.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

packet_source.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

packet_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

packet.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

packet.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

resample_input_audio_file.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

resample_input_audio_file.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

rtc_event_log_source.cc

Create RtcEventLogEncoderLegacy

2017-10-03 13:51:59 +00:00

rtc_event_log_source.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_analyze.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_encode.cc

Replacing the legacy tool RTPencode with a new rtp_encode

2017-11-24 09:05:48 +00:00

rtp_file_source.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_file_source.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

rtp_generator.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtp_generator.h

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

rtpcat.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00