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/*
* Copyright (c) 2016 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 "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/base/optional.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#ifdef WEBRTC_CODEC_G722
#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
#endif
#ifdef WEBRTC_CODEC_ILBC
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
#endif
#ifdef WEBRTC_CODEC_ISACFX
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h" // nogncheck
#endif
#ifdef WEBRTC_CODEC_ISAC
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h" // nogncheck
#endif
#ifdef WEBRTC_CODEC_OPUS
#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#endif
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
namespace webrtc {
namespace {
struct NamedDecoderConstructor {
const char* name;
// If |format| is good, return true and (if |out| isn't null) reset |*out| to
// a new decoder object. If the |format| is not good, return false.
bool (*constructor)(const SdpAudioFormat& format,
std::unique_ptr<AudioDecoder>* out);
};
// TODO(kwiberg): These factory functions should probably be moved to each
// decoder.
NamedDecoderConstructor decoder_constructors[] = {
{"pcmu",
[](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
if (format.clockrate_hz == 8000 && format.num_channels >= 1) {
if (out) {
out->reset(new AudioDecoderPcmU(format.num_channels));
}
return true;
} else {
return false;
}
}},
{"pcma",
[](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
if (format.clockrate_hz == 8000 && format.num_channels >= 1) {
if (out) {
out->reset(new AudioDecoderPcmA(format.num_channels));
}
return true;
} else {
return false;
}
}},
#ifdef WEBRTC_CODEC_ILBC
{"ilbc",
[](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
if (format.clockrate_hz == 8000 && format.num_channels == 1) {
if (out) {
out->reset(new AudioDecoderIlbc);
}
return true;
} else {
return false;
}
}},
#endif
#if defined(WEBRTC_CODEC_ISACFX)
{"isac",
[](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
if (format.clockrate_hz == 16000 && format.num_channels == 1) {
if (out) {
out->reset(new AudioDecoderIsacFix(format.clockrate_hz));
}
return true;
} else {
return false;
}
}},
#elif defined(WEBRTC_CODEC_ISAC)
{"isac",
[](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
if ((format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
format.num_channels == 1) {
if (out) {
out->reset(new AudioDecoderIsac(format.clockrate_hz));
}
return true;
} else {
return false;
}
}},
#endif
{"l16",
[](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
if (format.num_channels >= 1) {
if (out) {
out->reset(new AudioDecoderPcm16B(format.clockrate_hz,
format.num_channels));
}
return true;
} else {
return false;
}
}},
#ifdef WEBRTC_CODEC_G722
{"g722",
[](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
if (format.clockrate_hz == 8000) {
if (format.num_channels == 1) {
if (out) {
out->reset(new AudioDecoderG722);
}
return true;
} else if (format.num_channels == 2) {
if (out) {
out->reset(new AudioDecoderG722Stereo);
}
return true;
}
}
return false;
}},
#endif
#ifdef WEBRTC_CODEC_OPUS
{"opus",
[](const SdpAudioFormat& format, std::unique_ptr<AudioDecoder>* out) {
const rtc::Optional<int> num_channels = [&] {
auto stereo = format.parameters.find("stereo");
if (stereo != format.parameters.end()) {
if (stereo->second == "0") {
return rtc::Optional<int>(1);
} else if (stereo->second == "1") {
return rtc::Optional<int>(2);
} else {
return rtc::Optional<int>(); // Bad stereo parameter.
}
}
return rtc::Optional<int>(1); // Default to mono.
}();
if (format.clockrate_hz == 48000 && format.num_channels == 2 &&
num_channels) {
if (out) {
out->reset(new AudioDecoderOpus(*num_channels));
}
return true;
} else {
return false;
}
}},
#endif
};
class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
public:
std::vector<AudioCodecSpec> GetSupportedDecoders() override {
// Although this looks a bit strange, it means specs need only be initalized
// once, and that that initialization is thread-safe.
static std::vector<AudioCodecSpec> specs =
[]{
std::vector<AudioCodecSpec> specs;
#ifdef WEBRTC_CODEC_OPUS
AudioCodecSpec opus({"opus", 48000, 2, {
{"minptime", "10"},
{"useinbandfec", "1"}
}});
opus.allow_comfort_noise = false;
opus.supports_network_adaption = true;
specs.push_back(opus);
#endif
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
specs.push_back(AudioCodecSpec({"isac", 16000, 1}));
#endif
#if (defined(WEBRTC_CODEC_ISAC))
specs.push_back(AudioCodecSpec({"isac", 32000, 1}));
#endif
#ifdef WEBRTC_CODEC_G722
specs.push_back(AudioCodecSpec({"G722", 8000, 1}));
#endif
#ifdef WEBRTC_CODEC_ILBC
specs.push_back(AudioCodecSpec({"iLBC", 8000, 1}));
#endif
specs.push_back(AudioCodecSpec({"PCMU", 8000, 1}));
specs.push_back(AudioCodecSpec({"PCMA", 8000, 1}));
return specs;
}();
return specs;
}
bool IsSupportedDecoder(const SdpAudioFormat& format) override {
for (const auto& dc : decoder_constructors) {
if (STR_CASE_CMP(format.name.c_str(), dc.name) == 0) {
return dc.constructor(format, nullptr);
}
}
return false;
}
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format) override {
for (const auto& dc : decoder_constructors) {
if (STR_CASE_CMP(format.name.c_str(), dc.name) == 0) {
std::unique_ptr<AudioDecoder> decoder;
bool ok = dc.constructor(format, &decoder);
RTC_DCHECK_EQ(ok, decoder != nullptr);
if (decoder) {
const int expected_sample_rate_hz =
STR_CASE_CMP(format.name.c_str(), "g722") == 0
? 2 * format.clockrate_hz
: format.clockrate_hz;
RTC_CHECK_EQ(expected_sample_rate_hz, decoder->SampleRateHz());
}
return decoder;
}
}
return nullptr;
}
};
} // namespace
rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
return rtc::scoped_refptr<AudioDecoderFactory>(
new rtc::RefCountedObject<BuiltinAudioDecoderFactory>);
}
} // namespace webrtc