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webrtc/audio
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Harald Alvestrand 1f206b841e Use ArrayView in the IncomingRtcpPacket function. 
The lowest level and some of the highest levels of this function are
already using ArrayView. Make this consistent throughout.
Use deprecation for the old API rather than deleting it, since upstream
may be using it.

Bug: webrtc:14870
Change-Id: If5e1a6e9802ecf7e8e3ec27befb5167ca9985517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291706
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39241}
2023-02-01 12:19:03 +00:00
..
test Delete unused Audio Bwe integration test. 2023-01-26 09:31:44 +00:00
utility Remove dependency on rtc_base_approved from most targets 2022-04-25 12:15:30 +00:00
voip Use ArrayView in the IncomingRtcpPacket function. 2023-02-01 12:19:03 +00:00
audio_level.cc Migrate audio/ to use webrtc::Mutex 2020-07-06 14:21:38 +00:00
audio_level.h Migrate audio/ to use webrtc::Mutex 2020-07-06 14:21:38 +00:00
audio_receive_stream.cc Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
audio_receive_stream.h Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
audio_receive_stream_unittest.cc Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
audio_send_stream.cc Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead 2022-11-30 20:19:36 +00:00
audio_send_stream.h Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead 2022-11-30 20:19:36 +00:00
audio_send_stream_tests.cc CallTest: migrate timeouts to TimeDelta. 2022-08-16 12:06:54 +00:00
audio_send_stream_unittest.cc pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +00:00
audio_state.cc Rewrite AudioState null poller to use TaskQueueBase interface 2022-08-16 13:16:24 +00:00
audio_state.h Rewrite AudioState null poller to use TaskQueueBase interface 2022-08-16 13:16:24 +00:00
audio_state_unittest.cc Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
audio_transport_impl.cc Make capture timestamp optional in ADM. 2023-01-23 17:29:06 +00:00
audio_transport_impl.h Make capture timestamp optional in ADM. 2023-01-23 17:29:06 +00:00
BUILD.gn Reset encoder when audio send stream is stopped. 2023-01-26 15:20:02 +00:00
channel_receive.cc Use ArrayView in the IncomingRtcpPacket function. 2023-02-01 12:19:03 +00:00
channel_receive.h audio: make packets lost a signed integer 2022-11-01 11:46:49 +00:00
channel_receive_frame_transformer_delegate.cc Add GetContributionSources to TransformableIncomingAudioFrame 2022-10-11 12:52:21 +00:00
channel_receive_frame_transformer_delegate.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
channel_receive_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
channel_receive_unittest.cc Add a basic unittest for webrtc::voe::ChannelReceive 2023-01-25 20:06:26 +00:00
channel_send.cc Use ArrayView in the IncomingRtcpPacket function. 2023-02-01 12:19:03 +00:00
channel_send.h [Stats] Expose totalPacketSendDelay for audio as well. 2022-10-27 10:33:16 +00:00
channel_send_frame_transformer_delegate.cc Update audio code to not use implicit T* --> scoped_refptr<T> conversion 2022-01-13 15:49:49 +00:00
channel_send_frame_transformer_delegate.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
channel_send_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
channel_send_unittest.cc Update RTP timestamp based on capture timestamp when audio send stream is resumed. 2023-01-27 15:46:32 +00:00
conversion.h Make header files self contained. 2022-10-08 08:38:36 +00:00
DEPS pc: Add asynchronous RtpSender::SetParameters() call 2022-11-15 15:31:40 +00:00
mock_voe_channel_proxy.h Implement RTCOutboundRtpStreamStats.targetBitrate for audio. 2021-11-12 09:24:34 +00:00
OWNERS Add alessiob@webrtc.org in audio/OWNERS 2022-09-09 07:33:11 +00:00
remix_resample.cc Reland "Rename FATAL() into RTC_FATAL()." 2020-11-18 20:49:08 +00:00
remix_resample.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
remix_resample_unittest.cc Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00