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/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "api/audio_codecs/g711/audio_encoder_g711.h"
#include <memory>
#include <vector>
#include "absl/strings/match.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {
absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
const SdpAudioFormat& format) {
const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
(is_pcmu || is_pcma)) {
Config config;
config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
config.frame_size_ms = 20;
auto ptime_iter = format.parameters.find("ptime");
if (ptime_iter != format.parameters.end()) {
const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
if (ptime && *ptime > 0) {
config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
}
}
RTC_DCHECK(config.IsOk());
return config;
} else {
return absl::nullopt;
}
}
void AudioEncoderG711::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
for (const char* type : {"PCMU", "PCMA"}) {
specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
}
}
AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) {
RTC_DCHECK(config.IsOk());
return {8000, rtc::dchecked_cast<size_t>(config.num_channels),
64000 * config.num_channels};
}
std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
const Config& config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
RTC_DCHECK(config.IsOk());
switch (config.type) {
case Config::Type::kPcmU: {
AudioEncoderPcmU::Config impl_config;
impl_config.num_channels = config.num_channels;
impl_config.frame_size_ms = config.frame_size_ms;
impl_config.payload_type = payload_type;
return std::make_unique<AudioEncoderPcmU>(impl_config);
}
case Config::Type::kPcmA: {
AudioEncoderPcmA::Config impl_config;
impl_config.num_channels = config.num_channels;
impl_config.frame_size_ms = config.frame_size_ms;
impl_config.payload_type = payload_type;
return std::make_unique<AudioEncoderPcmA>(impl_config);
}
default: {
return nullptr;
}
}
}
} // namespace webrtc
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