1bd66642c3
All users call SetTelephoneEventForwardToDecoder(true). Setting the flag to true on construction, enables deletion of those calls, followed by deletion of the flag itself. The unused getter method TelephoneEventForwardToDecoder() is deleted right away. Bug: webrtc:7135 Change-Id: I8c52c957b3f074be7ffc425b3588402d1e42b844 Reviewed-on: https://webrtc-review.googlesource.com/90402 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24141}
236 lines
8.8 KiB
C++
236 lines
8.8 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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TelephoneEventHandler(),
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telephone_event_forward_to_decoder_(true),
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telephone_event_payload_type_(-1),
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cng_nb_payload_type_(-1),
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cng_wb_payload_type_(-1),
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cng_swb_payload_type_(-1),
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cng_fb_payload_type_(-1) {}
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RTPReceiverAudio::~RTPReceiverAudio() = default;
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// Outband TelephoneEvent(DTMF) detection
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void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
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bool forward_to_decoder) {
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rtc::CritScope lock(&crit_sect_);
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telephone_event_forward_to_decoder_ = forward_to_decoder;
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}
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bool RTPReceiverAudio::TelephoneEventPayloadType(int8_t payload_type) const {
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rtc::CritScope lock(&crit_sect_);
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return telephone_event_payload_type_ == payload_type;
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}
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TelephoneEventHandler* RTPReceiverAudio::GetTelephoneEventHandler() {
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return this;
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}
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bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type) {
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rtc::CritScope lock(&crit_sect_);
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return payload_type == cng_nb_payload_type_ ||
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payload_type == cng_wb_payload_type_ ||
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payload_type == cng_swb_payload_type_ ||
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payload_type == cng_fb_payload_type_;
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}
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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::OnNewPayloadTypeCreated(
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int payload_type,
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const SdpAudioFormat& audio_format) {
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rtc::CritScope lock(&crit_sect_);
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if (RtpUtility::StringCompare(audio_format.name.c_str(), "telephone-event",
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15)) {
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telephone_event_payload_type_ = payload_type;
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}
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if (RtpUtility::StringCompare(audio_format.name.c_str(), "cn", 2)) {
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// We support comfort noise at four different frequencies.
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if (audio_format.clockrate_hz == 8000) {
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cng_nb_payload_type_ = payload_type;
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} else if (audio_format.clockrate_hz == 16000) {
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cng_wb_payload_type_ = payload_type;
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} else if (audio_format.clockrate_hz == 32000) {
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cng_swb_payload_type_ = payload_type;
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} else if (audio_format.clockrate_hz == 48000) {
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cng_fb_payload_type_ = payload_type;
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} else {
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assert(false);
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return -1;
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}
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}
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return 0;
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}
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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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RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive(
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uint16_t last_payload_length) const {
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// Our CNG is 9 bytes; if it's a likely CNG the receiver needs to check
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// kRtpNoRtp against NetEq speech_type kOutputPLCtoCNG.
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if (last_payload_length < 10) { // our CNG is 9 bytes
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return kRtpNoRtp;
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} else {
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return kRtpDead;
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}
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}
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void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
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PayloadUnion* /* specific_payload */,
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bool* should_discard_changes) {
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*should_discard_changes =
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TelephoneEventPayloadType(payload_type) || CNGPayloadType(payload_type);
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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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bool telephone_event_packet =
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TelephoneEventPayloadType(rtp_header->header.payloadType);
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if (telephone_event_packet) {
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rtc::CritScope lock(&crit_sect_);
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// RFC 4733 2.3
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// 0 1 2 3
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// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// | event |E|R| volume | duration |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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//
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if (payload_data_length % 4 != 0) {
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return -1;
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}
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size_t number_of_events = payload_data_length / 4;
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// sanity
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if (number_of_events >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS) {
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number_of_events = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS;
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}
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for (size_t n = 0; n < number_of_events; ++n) {
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RTC_DCHECK_GE(payload_data_length, (4 * n) + 2);
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bool end = (payload_data[(4 * n) + 1] & 0x80) ? true : false;
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std::set<uint8_t>::iterator event =
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telephone_event_reported_.find(payload_data[4 * n]);
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if (event != telephone_event_reported_.end()) {
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// we have already seen this event
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if (end) {
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telephone_event_reported_.erase(payload_data[4 * n]);
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}
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} else {
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if (end) {
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// don't add if it's a end of a tone
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} else {
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telephone_event_reported_.insert(payload_data[4 * n]);
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}
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}
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}
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// RFC 4733 2.5.1.3 & 2.5.2.3 Long-Duration Events
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// should not be a problem since we don't care about the duration
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// RFC 4733 See 2.5.1.5. & 2.5.2.4. Multiple Events in a Packet
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}
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{
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rtc::CritScope lock(&crit_sect_);
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// check if it's a DTMF event, hence something we can playout
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if (telephone_event_packet) {
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if (!telephone_event_forward_to_decoder_) {
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// don't forward event to decoder
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return 0;
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}
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std::set<uint8_t>::iterator first = telephone_event_reported_.begin();
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if (first != telephone_event_reported_.end() && *first > 15) {
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// don't forward non DTMF events
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return 0;
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}
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}
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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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