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3ed46bd83b
Bug: webrtc:7135 Change-Id: Ic20d98cbfb8154f5abbc2501cbcccb950148e732 Reviewed-on: https://webrtc-review.googlesource.com/92396 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24219}
95 lines
4.1 KiB
C++
95 lines
4.1 KiB
C++
/*
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* Copyright (c) 2012 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/rtp_rtcp/source/rtp_receiver_audio.h"
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#include <assert.h> // assert
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#include <math.h> // pow()
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#include <string.h> // memcpy()
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#include "common_types.h" // NOLINT(build/include)
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#include "rtc_base/logging.h"
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#include "rtc_base/trace_event.h"
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namespace webrtc {
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RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
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RtpData* data_callback) {
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return new RTPReceiverAudio(data_callback);
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}
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RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
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: RTPReceiverStrategy(data_callback) {}
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RTPReceiverAudio::~RTPReceiverAudio() = default;
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// - Sample based or frame based codecs based on RFC 3551
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// -
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// - NOTE! There is one error in the RFC, stating G.722 uses 8 bits/samples.
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// - The correct rate is 4 bits/sample.
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// -
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// - name of sampling default
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// - encoding sample/frame bits/sample rate ms/frame ms/packet
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// -
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// - Sample based audio codecs
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// - DVI4 sample 4 var. 20
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// - G722 sample 4 16,000 20
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// - G726-40 sample 5 8,000 20
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// - G726-32 sample 4 8,000 20
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// - G726-24 sample 3 8,000 20
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// - G726-16 sample 2 8,000 20
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// - L8 sample 8 var. 20
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// - L16 sample 16 var. 20
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// - PCMA sample 8 var. 20
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// - PCMU sample 8 var. 20
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// -
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// - Frame based audio codecs
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// - G723 frame N/A 8,000 30 30
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// - G728 frame N/A 8,000 2.5 20
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// - G729 frame N/A 8,000 10 20
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// - G729D frame N/A 8,000 10 20
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// - G729E frame N/A 8,000 10 20
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// - GSM frame N/A 8,000 20 20
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// - GSM-EFR frame N/A 8,000 20 20
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// - LPC frame N/A 8,000 20 20
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// - MPA frame N/A var. var.
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// -
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// - G7221 frame N/A
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int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
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const PayloadUnion& specific_payload,
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const uint8_t* payload,
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size_t payload_length,
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int64_t timestamp_ms) {
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if (first_packet_received_()) {
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RTC_LOG(LS_INFO) << "Received first audio RTP packet";
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}
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return ParseAudioCodecSpecific(rtp_header, payload, payload_length,
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specific_payload.audio_payload());
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}
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// We are not allowed to have any critsects when calling data_callback.
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int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
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WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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size_t payload_length,
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const AudioPayload& audio_specific) {
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RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
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const size_t payload_data_length =
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payload_length - rtp_header->header.paddingLength;
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if (payload_data_length == 0) {
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rtp_header->frameType = kEmptyFrame;
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return data_callback_->OnReceivedPayloadData(nullptr, 0, rtp_header);
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}
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return data_callback_->OnReceivedPayloadData(payload_data,
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payload_data_length, rtp_header);
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}
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} // namespace webrtc
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