webrtc/audio
Daniel Lee 63658d06ec Revert "Ensure that we always set values for min and max audio bitrate."
This reverts commit e47aee3b86.

Reason for revert: Breaks downstream project

Original change's description:
> Ensure that we always set values for min and max audio bitrate.
> 
> Use (in order from lowest to highest precedence):
> -- fixed 32000bps
> -- fixed target bitrate from codec
> -- explicit values from the rtp encoding parameters
> -- Final precedence is given to field trial values from
>    WebRTC-Audio-Allocation
> 
> Bug: webrtc:10487
> Change-Id: I7e289f209a927785572058b6fbfdf60fa14edf05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126229
> Reviewed-by: Minyue Li <minyue@google.com>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Daniel Lee <dklee@google.com>
> Cr-Commit-Position: refs/heads/master@{#27667}

TBR=solenberg@webrtc.org,stefan@webrtc.org,srte@webrtc.org,crodbro@webrtc.org,minyue@webrtc.org,minyue@google.com,dklee@google.com

Change-Id: Ie975cf40e65105d1e4cfab417b220b6bfc34592b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10487
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133481
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Daniel Lee <dklee@google.com>
Cr-Commit-Position: refs/heads/master@{#27670}
2019-04-17 15:47:00 +00:00
..
test Add base class NetworkPredictor and NetworkPredictorFactory and wire up. 2019-04-10 12:38:58 +00:00
utility Implicitly suppress //build/config/clang:find_bad_constructs. 2019-03-01 10:18:17 +00:00
audio_level.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_level.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_receive_stream.cc Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 2019-04-15 16:06:01 +00:00
audio_receive_stream.h Delete a few return values from audio streams and video send streams. 2019-03-06 10:56:08 +00:00
audio_receive_stream_unittest.cc Fully qualify googletest symbols. 2019-04-09 17:18:20 +00:00
audio_send_stream.cc Revert "Ensure that we always set values for min and max audio bitrate." 2019-04-17 15:47:00 +00:00
audio_send_stream.h Revert "Ensure that we always set values for min and max audio bitrate." 2019-04-17 15:47:00 +00:00
audio_send_stream_tests.cc Reland "Delete test/constants.h" 2019-02-19 08:51:20 +00:00
audio_send_stream_unittest.cc Revert "Ensure that we always set values for min and max audio bitrate." 2019-04-17 15:47:00 +00:00
audio_state.cc Deprecating ThreadChecker specific interface. 2019-04-08 16:58:07 +00:00
audio_state.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_state_unittest.cc Fully qualify googletest symbols. 2019-04-09 17:18:20 +00:00
audio_transport_impl.cc Fixes ClangTidy errors in audio/ 2019-03-15 01:55:52 +00:00
audio_transport_impl.h Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
BUILD.gn Avoid using global task queue factory in audio/ unittests 2019-03-22 15:53:28 +00:00
channel_receive.cc Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 2019-04-15 16:06:01 +00:00
channel_receive.h Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 2019-04-15 16:06:01 +00:00
channel_send.cc Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. 2019-04-17 13:04:50 +00:00
channel_send.h Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. 2019-04-17 13:04:50 +00:00
conversion.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
DEPS Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
mock_voe_channel_proxy.h Move ownership of RTPSenderAudio to ChannelSend. 2019-03-06 17:15:00 +00:00
null_audio_poller.cc Deprecating ThreadChecker specific interface. 2019-04-08 16:58:07 +00:00
null_audio_poller.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
remix_resample.cc Receive-side ready for multiple channels. 2019-01-29 12:43:23 +00:00
remix_resample.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
remix_resample_unittest.cc Qualify cmath functions. 2019-03-19 09:48:44 +00:00
transport_feedback_packet_loss_tracker.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
transport_feedback_packet_loss_tracker.h Remove clang:find_bad_constructs suppression from call:call. 2018-08-29 11:57:00 +00:00
transport_feedback_packet_loss_tracker_unittest.cc Fully qualify googletest symbols. 2019-04-09 17:18:20 +00:00