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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/opus/audio_decoder_opus.h"
#include <map>
#include <memory>
#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
int GetDefaultNumChannels(const FieldTrialsView& field_trials) {
return field_trials.IsEnabled("WebRTC-Audio-OpusDecodeStereoByDefault") ? 2
: 1;
}
} // namespace
bool AudioDecoderOpus::Config::IsOk() const {
if (sample_rate_hz != 16000 && sample_rate_hz != 48000) {
// Unsupported sample rate. (libopus supports a few other rates as
// well; we can add support for them when needed.)
return false;
}
return !num_channels.has_value() || *num_channels == 1 || *num_channels == 2;
}
std::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "opus") ||
format.clockrate_hz != 48000 || format.num_channels != 2) {
return std::nullopt;
}
Config config;
// Parse the "stereo" codec parameter. If set, it overrides the default number
// of channels.
const auto stereo_param = format.parameters.find("stereo");
if (stereo_param != format.parameters.end()) {
if (stereo_param->second == "0") {
config.num_channels = 1;
} else if (stereo_param->second == "1") {
config.num_channels = 2;
} else {
// Malformed stereo parameter.
return std::nullopt;
}
}
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return std::nullopt;
}
return config;
}
void AudioDecoderOpus::AppendSupportedDecoders(
std::vector<AudioCodecSpec>* specs) {
AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
opus_info.allow_comfort_noise = false;
opus_info.supports_network_adaption = true;
SdpAudioFormat opus_format(
{"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}});
specs->push_back({std::move(opus_format), opus_info});
}
std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
const Environment& env,
Config config) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;
}
return std::make_unique<AudioDecoderOpusImpl>(
env.field_trials(),
config.num_channels.value_or(GetDefaultNumChannels(env.field_trials())),
config.sample_rate_hz);
}
} // namespace webrtc
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