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https://github.com/mollyim/webrtc.git
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08973eed36
WebRTC internal code should always used include paths that starts from the root of the project and that clearly identify the header file. This allows 'gn check' to actually keep dependencies under control because 'gn check' cannot enforce anything if the include path is not fully qualified (starting from the root of the project). Bug: webrtc:8815 Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc Reviewed-on: https://webrtc-review.googlesource.com/46101 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21847}
89 lines
2.9 KiB
C
89 lines
2.9 KiB
C
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/codecs/isac/main/source/structs.h"
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#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h"
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#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h"
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#include "modules/audio_coding/codecs/isac/main/source/codec.h"
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int
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WebRtcIsac_EstimateBandwidth(
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BwEstimatorstr* bwest_str,
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Bitstr* streamdata,
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size_t packet_size,
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uint16_t rtp_seq_number,
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uint32_t send_ts,
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uint32_t arr_ts,
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enum IsacSamplingRate encoderSampRate,
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enum IsacSamplingRate decoderSampRate)
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{
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int16_t index;
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int16_t frame_samples;
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uint32_t sendTimestampIn16kHz;
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uint32_t arrivalTimestampIn16kHz;
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uint32_t diffSendTime;
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uint32_t diffArrivalTime;
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int err;
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/* decode framelength and BW estimation */
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err = WebRtcIsac_DecodeFrameLen(streamdata, &frame_samples);
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if(err < 0) // error check
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{
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return err;
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}
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err = WebRtcIsac_DecodeSendBW(streamdata, &index);
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if(err < 0) // error check
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{
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return err;
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}
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/* UPDATE ESTIMATES FROM OTHER SIDE */
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err = WebRtcIsac_UpdateUplinkBwImpl(bwest_str, index, encoderSampRate);
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if(err < 0)
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{
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return err;
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}
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// We like BWE to work at 16 kHz sampling rate,
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// therefore, we have to change the timestamps accordingly.
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// translate the send timestamp if required
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diffSendTime = (uint32_t)((uint32_t)send_ts -
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(uint32_t)bwest_str->senderTimestamp);
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bwest_str->senderTimestamp = send_ts;
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diffArrivalTime = (uint32_t)((uint32_t)arr_ts -
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(uint32_t)bwest_str->receiverTimestamp);
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bwest_str->receiverTimestamp = arr_ts;
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if(decoderSampRate == kIsacSuperWideband)
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{
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diffArrivalTime = (uint32_t)diffArrivalTime >> 1;
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diffSendTime = (uint32_t)diffSendTime >> 1;
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}
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// arrival timestamp in 16 kHz
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arrivalTimestampIn16kHz = (uint32_t)((uint32_t)
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bwest_str->prev_rec_arr_ts + (uint32_t)diffArrivalTime);
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// send timestamp in 16 kHz
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sendTimestampIn16kHz = (uint32_t)((uint32_t)
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bwest_str->prev_rec_send_ts + (uint32_t)diffSendTime);
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err = WebRtcIsac_UpdateBandwidthEstimator(bwest_str, rtp_seq_number,
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(frame_samples * 1000) / FS, sendTimestampIn16kHz,
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arrivalTimestampIn16kHz, packet_size);
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// error check
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if(err < 0)
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{
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return err;
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}
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return 0;
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}
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