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webrtc/modules/audio_coding/neteq/tools
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Henrik Lundin 9f2e624024 Break out NetEqEventLogInput to separate source files 
Building NetEqEventLogInput requires protobuf support, while building
NetEqRtpDumpInput located in the same file does not. This makes both
classes unusable when protobuf support is lacking. With this CL, the
NetEqEventLogInput is broken out into separate files, to allow usage
of NetEqRtpDumpInput even when protobufs are not supported.

Bug: webrtc:9421
Change-Id: I55efec4ec259713654566cdaa00d2e34c5e9a60f
Reviewed-on: https://webrtc-review.googlesource.com/84587
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23803}
2018-07-02 14:15:29 +00:00
..
audio_checksum.h Use generic MessageDigest class instead of MD5 / SHA-1 specific classes. 2017-12-21 12:39:50 +00:00
audio_loop.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_loop.h Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
encode_neteq_input.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
fake_decode_from_file.cc Implement PacketDuration() for FakeDecoderFromFile. 2018-06-29 08:32:36 +00:00
fake_decode_from_file.h Implement PacketDuration() for FakeDecoderFromFile. 2018-06-29 08:32:36 +00:00
input_audio_file.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
input_audio_file.h Reformat the WebRTC code base 2018-06-19 14:00:39 +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
neteq_delay_analyzer.cc Aligning time in audio jitter buffer plot to other plots in rtc event log visualizer. 2018-06-21 14:23:53 +00:00
neteq_delay_analyzer.h Aligning time in audio jitter buffer plot to other plots in rtc event log visualizer. 2018-06-21 14:23:53 +00:00
neteq_event_log_input.cc Break out NetEqEventLogInput to separate source files 2018-07-02 14:15:29 +00:00
neteq_event_log_input.h Break out NetEqEventLogInput to separate source files 2018-07-02 14:15:29 +00:00
neteq_external_decoder_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
neteq_external_decoder_test.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
neteq_input.cc Reland "NetEq: Deprecate playout modes Fax, Off and Streaming" 2018-07-02 10:20:33 +00:00
neteq_input.h Reland "NetEq: Deprecate playout modes Fax, Off and Streaming" 2018-07-02 10:20:33 +00:00
neteq_packet_source_input.cc Break out NetEqEventLogInput to separate source files 2018-07-02 14:15:29 +00:00
neteq_packet_source_input.h Break out NetEqEventLogInput to separate source files 2018-07-02 14:15:29 +00:00
neteq_performance_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
neteq_quality_test.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
neteq_replacement_input.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
neteq_replacement_input.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
neteq_rtpplay.cc Break out NetEqEventLogInput to separate source files 2018-07-02 14:15:29 +00:00
neteq_stats_getter.cc Adding NetEq lifetime stats to event log visualizer. 2018-06-26 11:27:09 +00:00
neteq_stats_getter.h Adding NetEq lifetime stats to event log visualizer. 2018-06-26 11:27:09 +00:00
neteq_test.cc Reland "NetEq: Deprecate playout modes Fax, Off and Streaming" 2018-07-02 10:20:33 +00:00
neteq_test.h Reland "NetEq: Deprecate playout modes Fax, Off and Streaming" 2018-07-02 10:20:33 +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.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
packet.h Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Split LoggedBweProbeResult into -Success and -Failure. 2018-05-29 13:41:04 +00:00
rtc_event_log_source.h Reland "Create new API for RtcEventLogParser." 2018-04-27 14:46:51 +00:00
rtp_analyze.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_encode.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_file_source.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_generator.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtp_jitter.cc Replacing the legacy tool RTPjitter with a new rtp_jitter 2017-11-24 13:38:59 +00:00
rtpcat.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00