f1b08da5b4
NetEqDecoder is still used in the external interfaces, but this change opens up the ability to use SdpAudioFormats directly, once appropriate interfaces have been added. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2355503002 Cr-Commit-Position: refs/heads/master@{#14368}
381 lines
14 KiB
C++
381 lines
14 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/rent_a_codec.h"
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#include <memory>
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#include <utility>
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#include "webrtc/base/logging.h"
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#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
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#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
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#ifdef WEBRTC_CODEC_G722
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#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
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#endif
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#ifdef WEBRTC_CODEC_ILBC
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#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
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#endif
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#ifdef WEBRTC_CODEC_ISACFX
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#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
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#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
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#endif
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#ifdef WEBRTC_CODEC_ISAC
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#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
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#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
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#endif
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#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
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#ifdef WEBRTC_CODEC_RED
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#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
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#endif
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#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h"
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#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
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#if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
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#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
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#endif
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namespace webrtc {
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namespace acm2 {
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rtc::Optional<SdpAudioFormat> RentACodec::NetEqDecoderToSdpAudioFormat(
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NetEqDecoder nd) {
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switch (nd) {
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case NetEqDecoder::kDecoderPCMu:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcmu", 8000, 1));
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case NetEqDecoder::kDecoderPCMa:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcma", 8000, 1));
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case NetEqDecoder::kDecoderPCMu_2ch:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcmu", 8000, 2));
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case NetEqDecoder::kDecoderPCMa_2ch:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("pcma", 8000, 2));
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case NetEqDecoder::kDecoderILBC:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("ilbc", 8000, 1));
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case NetEqDecoder::kDecoderISAC:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("isac", 16000, 1));
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case NetEqDecoder::kDecoderISACswb:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("isac", 32000, 1));
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case NetEqDecoder::kDecoderPCM16B:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 8000, 1));
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case NetEqDecoder::kDecoderPCM16Bwb:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 16000, 1));
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case NetEqDecoder::kDecoderPCM16Bswb32kHz:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 32000, 1));
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case NetEqDecoder::kDecoderPCM16Bswb48kHz:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 48000, 1));
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case NetEqDecoder::kDecoderPCM16B_2ch:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 8000, 2));
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case NetEqDecoder::kDecoderPCM16Bwb_2ch:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 16000, 2));
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case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 32000, 2));
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case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 48000, 2));
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case NetEqDecoder::kDecoderPCM16B_5ch:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("l16", 8000, 5));
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case NetEqDecoder::kDecoderG722:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("g722", 8000, 1));
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case NetEqDecoder::kDecoderG722_2ch:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("g722", 8000, 2));
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case NetEqDecoder::kDecoderOpus:
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return rtc::Optional<SdpAudioFormat>(
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SdpAudioFormat("opus", 48000, 2,
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std::map<std::string, std::string>{{"stereo", "0"}}));
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case NetEqDecoder::kDecoderOpus_2ch:
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return rtc::Optional<SdpAudioFormat>(
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SdpAudioFormat("opus", 48000, 2,
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std::map<std::string, std::string>{{"stereo", "1"}}));
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case NetEqDecoder::kDecoderRED:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("red", 8000, 1));
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case NetEqDecoder::kDecoderAVT:
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return rtc::Optional<SdpAudioFormat>(
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SdpAudioFormat("telephone-event", 8000, 1));
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case NetEqDecoder::kDecoderCNGnb:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("cn", 8000, 1));
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case NetEqDecoder::kDecoderCNGwb:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("cn", 16000, 1));
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case NetEqDecoder::kDecoderCNGswb32kHz:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("cn", 32000, 1));
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case NetEqDecoder::kDecoderCNGswb48kHz:
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return rtc::Optional<SdpAudioFormat>(SdpAudioFormat("cn", 48000, 1));
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default:
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return rtc::Optional<SdpAudioFormat>();
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}
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}
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rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
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const char* payload_name,
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int sampling_freq_hz,
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size_t channels) {
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return CodecIdFromIndex(
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ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels));
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}
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rtc::Optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
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rtc::Optional<int> mi = CodecIndexFromId(codec_id);
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return mi ? rtc::Optional<CodecInst>(Database()[*mi])
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: rtc::Optional<CodecInst>();
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}
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rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
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const CodecInst& codec_inst) {
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return CodecIdFromIndex(ACMCodecDB::CodecNumber(codec_inst));
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}
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rtc::Optional<CodecInst> RentACodec::CodecInstByParams(const char* payload_name,
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int sampling_freq_hz,
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size_t channels) {
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rtc::Optional<CodecId> codec_id =
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CodecIdByParams(payload_name, sampling_freq_hz, channels);
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if (!codec_id)
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return rtc::Optional<CodecInst>();
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rtc::Optional<CodecInst> ci = CodecInstById(*codec_id);
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RTC_DCHECK(ci);
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// Keep the number of channels from the function call. For most codecs it
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// will be the same value as in default codec settings, but not for all.
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ci->channels = channels;
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return ci;
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}
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bool RentACodec::IsCodecValid(const CodecInst& codec_inst) {
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return ACMCodecDB::CodecNumber(codec_inst) >= 0;
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}
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rtc::Optional<bool> RentACodec::IsSupportedNumChannels(CodecId codec_id,
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size_t num_channels) {
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auto i = CodecIndexFromId(codec_id);
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return i ? rtc::Optional<bool>(
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ACMCodecDB::codec_settings_[*i].channel_support >=
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num_channels)
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: rtc::Optional<bool>();
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}
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rtc::ArrayView<const CodecInst> RentACodec::Database() {
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return rtc::ArrayView<const CodecInst>(ACMCodecDB::database_,
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NumberOfCodecs());
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}
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rtc::Optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
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CodecId codec_id,
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size_t num_channels) {
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rtc::Optional<int> i = CodecIndexFromId(codec_id);
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if (!i)
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return rtc::Optional<NetEqDecoder>();
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const NetEqDecoder ned = ACMCodecDB::neteq_decoders_[*i];
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return rtc::Optional<NetEqDecoder>(
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(ned == NetEqDecoder::kDecoderOpus && num_channels == 2)
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? NetEqDecoder::kDecoderOpus_2ch
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: ned);
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}
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RentACodec::RegistrationResult RentACodec::RegisterCngPayloadType(
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std::map<int, int>* pt_map,
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const CodecInst& codec_inst) {
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if (STR_CASE_CMP(codec_inst.plname, "CN") != 0)
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return RegistrationResult::kSkip;
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switch (codec_inst.plfreq) {
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case 8000:
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case 16000:
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case 32000:
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case 48000:
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(*pt_map)[codec_inst.plfreq] = codec_inst.pltype;
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return RegistrationResult::kOk;
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default:
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return RegistrationResult::kBadFreq;
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}
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}
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RentACodec::RegistrationResult RentACodec::RegisterRedPayloadType(
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std::map<int, int>* pt_map,
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const CodecInst& codec_inst) {
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if (STR_CASE_CMP(codec_inst.plname, "RED") != 0)
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return RegistrationResult::kSkip;
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switch (codec_inst.plfreq) {
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case 8000:
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(*pt_map)[codec_inst.plfreq] = codec_inst.pltype;
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return RegistrationResult::kOk;
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default:
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return RegistrationResult::kBadFreq;
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}
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}
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namespace {
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// Returns a new speech encoder, or null on error.
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// TODO(kwiberg): Don't handle errors here (bug 5033)
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std::unique_ptr<AudioEncoder> CreateEncoder(
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const CodecInst& speech_inst,
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const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
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#if defined(WEBRTC_CODEC_ISACFX)
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if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
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return std::unique_ptr<AudioEncoder>(
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new AudioEncoderIsacFix(speech_inst, bwinfo));
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#endif
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#if defined(WEBRTC_CODEC_ISAC)
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if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
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return std::unique_ptr<AudioEncoder>(
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new AudioEncoderIsac(speech_inst, bwinfo));
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#endif
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#ifdef WEBRTC_CODEC_OPUS
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if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
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return std::unique_ptr<AudioEncoder>(new AudioEncoderOpus(speech_inst));
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#endif
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if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
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return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmU(speech_inst));
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if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
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return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmA(speech_inst));
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if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
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return std::unique_ptr<AudioEncoder>(new AudioEncoderPcm16B(speech_inst));
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#ifdef WEBRTC_CODEC_ILBC
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if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
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return std::unique_ptr<AudioEncoder>(new AudioEncoderIlbc(speech_inst));
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#endif
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#ifdef WEBRTC_CODEC_G722
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if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
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return std::unique_ptr<AudioEncoder>(new AudioEncoderG722(speech_inst));
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#endif
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LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname;
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return std::unique_ptr<AudioEncoder>();
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}
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std::unique_ptr<AudioEncoder> CreateRedEncoder(
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std::unique_ptr<AudioEncoder> encoder,
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int red_payload_type) {
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#ifdef WEBRTC_CODEC_RED
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AudioEncoderCopyRed::Config config;
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config.payload_type = red_payload_type;
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config.speech_encoder = std::move(encoder);
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return std::unique_ptr<AudioEncoder>(
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new AudioEncoderCopyRed(std::move(config)));
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#else
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return std::unique_ptr<AudioEncoder>();
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#endif
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}
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std::unique_ptr<AudioEncoder> CreateCngEncoder(
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std::unique_ptr<AudioEncoder> encoder,
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int payload_type,
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ACMVADMode vad_mode) {
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AudioEncoderCng::Config config;
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config.num_channels = encoder->NumChannels();
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config.payload_type = payload_type;
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config.speech_encoder = std::move(encoder);
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switch (vad_mode) {
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case VADNormal:
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config.vad_mode = Vad::kVadNormal;
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break;
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case VADLowBitrate:
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config.vad_mode = Vad::kVadLowBitrate;
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break;
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case VADAggr:
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config.vad_mode = Vad::kVadAggressive;
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break;
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case VADVeryAggr:
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config.vad_mode = Vad::kVadVeryAggressive;
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break;
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default:
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FATAL();
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}
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return std::unique_ptr<AudioEncoder>(new AudioEncoderCng(std::move(config)));
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}
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std::unique_ptr<AudioDecoder> CreateIsacDecoder(
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int sample_rate_hz,
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const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
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#if defined(WEBRTC_CODEC_ISACFX)
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return std::unique_ptr<AudioDecoder>(
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new AudioDecoderIsacFix(sample_rate_hz, bwinfo));
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#elif defined(WEBRTC_CODEC_ISAC)
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return std::unique_ptr<AudioDecoder>(
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new AudioDecoderIsac(sample_rate_hz, bwinfo));
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#else
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FATAL() << "iSAC is not supported.";
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return std::unique_ptr<AudioDecoder>();
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#endif
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}
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} // namespace
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RentACodec::RentACodec() {
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#if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
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isac_bandwidth_info_ = new LockedIsacBandwidthInfo;
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#endif
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}
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RentACodec::~RentACodec() = default;
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std::unique_ptr<AudioEncoder> RentACodec::RentEncoder(
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const CodecInst& codec_inst) {
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return CreateEncoder(codec_inst, isac_bandwidth_info_);
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}
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RentACodec::StackParameters::StackParameters() {
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// Register the default payload types for RED and CNG.
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for (const CodecInst& ci : RentACodec::Database()) {
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RentACodec::RegisterCngPayloadType(&cng_payload_types, ci);
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RentACodec::RegisterRedPayloadType(&red_payload_types, ci);
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}
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}
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RentACodec::StackParameters::~StackParameters() = default;
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std::unique_ptr<AudioEncoder> RentACodec::RentEncoderStack(
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StackParameters* param) {
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if (!param->speech_encoder)
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return nullptr;
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if (param->use_codec_fec) {
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// Switch FEC on. On failure, remember that FEC is off.
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if (!param->speech_encoder->SetFec(true))
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param->use_codec_fec = false;
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} else {
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// Switch FEC off. This shouldn't fail.
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const bool success = param->speech_encoder->SetFec(false);
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RTC_DCHECK(success);
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}
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auto pt = [¶m](const std::map<int, int>& m) {
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auto it = m.find(param->speech_encoder->SampleRateHz());
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return it == m.end() ? rtc::Optional<int>()
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: rtc::Optional<int>(it->second);
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};
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auto cng_pt = pt(param->cng_payload_types);
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param->use_cng =
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param->use_cng && cng_pt && param->speech_encoder->NumChannels() == 1;
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auto red_pt = pt(param->red_payload_types);
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param->use_red = param->use_red && red_pt;
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if (param->use_cng || param->use_red) {
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// The RED and CNG encoders need to be in sync with the speech encoder, so
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// reset the latter to ensure its buffer is empty.
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param->speech_encoder->Reset();
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}
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std::unique_ptr<AudioEncoder> encoder_stack =
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std::move(param->speech_encoder);
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if (param->use_red) {
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encoder_stack = CreateRedEncoder(std::move(encoder_stack), *red_pt);
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}
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if (param->use_cng) {
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encoder_stack =
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CreateCngEncoder(std::move(encoder_stack), *cng_pt, param->vad_mode);
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}
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return encoder_stack;
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}
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std::unique_ptr<AudioDecoder> RentACodec::RentIsacDecoder(int sample_rate_hz) {
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return CreateIsacDecoder(sample_rate_hz, isac_bandwidth_info_);
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}
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} // namespace acm2
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} // namespace webrtc
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