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https://github.com/mollyim/webrtc.git
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e113fdf974
Signal only uses the Opus codec. Therefore, not adding the other codecs to the runtime list. Also disabling CN (comfort noise) since it is not needed for Opus and telephone-events, since there is no DTMF in the Signal clients. The patch simply comments out some code sections.
87 lines
2.9 KiB
C++
87 lines
2.9 KiB
C++
/*
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* Copyright (c) 2017 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 "api/audio_codecs/g711/audio_encoder_g711.h"
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#include <memory>
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#include <vector>
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#include "absl/strings/match.h"
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#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "rtc_base/string_to_number.h"
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namespace webrtc {
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absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
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const SdpAudioFormat& format) {
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const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
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const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
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if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
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(is_pcmu || is_pcma)) {
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Config config;
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config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
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config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
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config.frame_size_ms = 20;
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auto ptime_iter = format.parameters.find("ptime");
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if (ptime_iter != format.parameters.end()) {
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const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
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if (ptime && *ptime > 0) {
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config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
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}
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}
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RTC_DCHECK(config.IsOk());
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return config;
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} else {
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return absl::nullopt;
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}
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}
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void AudioEncoderG711::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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// for (const char* type : {"PCMU", "PCMA"}) {
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// specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
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// }
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}
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AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) {
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RTC_DCHECK(config.IsOk());
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return {8000, rtc::dchecked_cast<size_t>(config.num_channels),
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64000 * config.num_channels};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
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const Config& config,
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int payload_type,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
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RTC_DCHECK(config.IsOk());
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switch (config.type) {
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case Config::Type::kPcmU: {
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AudioEncoderPcmU::Config impl_config;
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impl_config.num_channels = config.num_channels;
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impl_config.frame_size_ms = config.frame_size_ms;
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impl_config.payload_type = payload_type;
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return std::make_unique<AudioEncoderPcmU>(impl_config);
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}
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case Config::Type::kPcmA: {
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AudioEncoderPcmA::Config impl_config;
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impl_config.num_channels = config.num_channels;
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impl_config.frame_size_ms = config.frame_size_ms;
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impl_config.payload_type = payload_type;
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return std::make_unique<AudioEncoderPcmA>(impl_config);
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}
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default: {
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return nullptr;
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}
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}
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}
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} // namespace webrtc
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