
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
195 lines
6.3 KiB
C++
195 lines
6.3 KiB
C++
/*
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* Copyright (c) 2015 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 "webrtc/modules/audio_coding/acm2/codec_manager.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/format_macros.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
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#include "webrtc/system_wrappers/include/trace.h"
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namespace webrtc {
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namespace acm2 {
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namespace {
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// Check if the given codec is a valid to be registered as send codec.
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int IsValidSendCodec(const CodecInst& send_codec) {
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int dummy_id = 0;
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if ((send_codec.channels != 1) && (send_codec.channels != 2)) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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"Wrong number of channels (%" PRIuS ", only mono and stereo "
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"are supported)",
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send_codec.channels);
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return -1;
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}
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auto maybe_codec_id = RentACodec::CodecIdByInst(send_codec);
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if (!maybe_codec_id) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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"Invalid codec setting for the send codec.");
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return -1;
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}
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// Telephone-event cannot be a send codec.
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if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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"telephone-event cannot be a send codec");
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return -1;
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}
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if (!RentACodec::IsSupportedNumChannels(*maybe_codec_id, send_codec.channels)
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.value_or(false)) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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"%" PRIuS " number of channels not supportedn for %s.",
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send_codec.channels, send_codec.plname);
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return -1;
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}
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return RentACodec::CodecIndexFromId(*maybe_codec_id).value_or(-1);
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}
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bool IsOpus(const CodecInst& codec) {
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return
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#ifdef WEBRTC_CODEC_OPUS
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!STR_CASE_CMP(codec.plname, "opus") ||
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#endif
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false;
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}
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} // namespace
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CodecManager::CodecManager() {
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thread_checker_.DetachFromThread();
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}
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CodecManager::~CodecManager() = default;
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bool CodecManager::RegisterEncoder(const CodecInst& send_codec) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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int codec_id = IsValidSendCodec(send_codec);
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// Check for reported errors from function IsValidSendCodec().
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if (codec_id < 0) {
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return false;
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}
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int dummy_id = 0;
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switch (RentACodec::RegisterRedPayloadType(
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&codec_stack_params_.red_payload_types, send_codec)) {
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case RentACodec::RegistrationResult::kOk:
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return true;
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case RentACodec::RegistrationResult::kBadFreq:
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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"RegisterSendCodec() failed, invalid frequency for RED"
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" registration");
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return false;
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case RentACodec::RegistrationResult::kSkip:
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break;
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}
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switch (RentACodec::RegisterCngPayloadType(
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&codec_stack_params_.cng_payload_types, send_codec)) {
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case RentACodec::RegistrationResult::kOk:
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return true;
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case RentACodec::RegistrationResult::kBadFreq:
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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"RegisterSendCodec() failed, invalid frequency for CNG"
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" registration");
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return false;
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case RentACodec::RegistrationResult::kSkip:
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break;
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}
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if (IsOpus(send_codec)) {
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// VAD/DTX not supported.
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codec_stack_params_.use_cng = false;
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}
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send_codec_inst_ = rtc::Optional<CodecInst>(send_codec);
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codec_stack_params_.speech_encoder = nullptr; // Caller must recreate it.
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return true;
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}
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CodecInst CodecManager::ForgeCodecInst(
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const AudioEncoder* external_speech_encoder) {
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CodecInst ci;
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ci.channels = external_speech_encoder->NumChannels();
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ci.plfreq = external_speech_encoder->SampleRateHz();
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ci.pacsize = rtc::CheckedDivExact(
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static_cast<int>(external_speech_encoder->Max10MsFramesInAPacket() *
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ci.plfreq),
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100);
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ci.pltype = -1; // Not valid.
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ci.rate = -1; // Not valid.
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static const char kName[] = "external";
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memcpy(ci.plname, kName, sizeof(kName));
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return ci;
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}
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bool CodecManager::SetCopyRed(bool enable) {
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if (enable && codec_stack_params_.use_codec_fec) {
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WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
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"Codec internal FEC and RED cannot be co-enabled.");
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return false;
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}
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if (enable && send_codec_inst_ &&
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codec_stack_params_.red_payload_types.count(send_codec_inst_->plfreq) <
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1) {
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WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
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"Cannot enable RED at %i Hz.", send_codec_inst_->plfreq);
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return false;
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}
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codec_stack_params_.use_red = enable;
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return true;
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}
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bool CodecManager::SetVAD(bool enable, ACMVADMode mode) {
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// Sanity check of the mode.
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RTC_DCHECK(mode == VADNormal || mode == VADLowBitrate || mode == VADAggr ||
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mode == VADVeryAggr);
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// Check that the send codec is mono. We don't support VAD/DTX for stereo
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// sending.
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const bool stereo_send =
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codec_stack_params_.speech_encoder
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? (codec_stack_params_.speech_encoder->NumChannels() != 1)
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: false;
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if (enable && stereo_send) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"VAD/DTX not supported for stereo sending");
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return false;
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}
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// TODO(kwiberg): This doesn't protect Opus when injected as an external
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// encoder.
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if (send_codec_inst_ && IsOpus(*send_codec_inst_)) {
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// VAD/DTX not supported, but don't fail.
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enable = false;
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}
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codec_stack_params_.use_cng = enable;
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codec_stack_params_.vad_mode = mode;
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return true;
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}
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bool CodecManager::SetCodecFEC(bool enable_codec_fec) {
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if (enable_codec_fec && codec_stack_params_.use_red) {
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WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
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"Codec internal FEC and RED cannot be co-enabled.");
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return false;
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}
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codec_stack_params_.use_codec_fec = enable_codec_fec;
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return true;
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}
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} // namespace acm2
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} // namespace webrtc
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