mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00

None of the built-in codecs do anything with the ID, but callers will soon require them to accept it. Bug: webrtc:8941 Change-Id: I0eb77db82d72c7d34cff639fecb67c1e6ec421bf Reviewed-on: https://webrtc-review.googlesource.com/58089 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22259}
86 lines
3 KiB
C++
86 lines
3 KiB
C++
/*
|
|
* 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 "common_types.h" // NOLINT(build/include)
|
|
#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/ptr_util.h"
|
|
#include "rtc_base/string_to_number.h"
|
|
|
|
namespace webrtc {
|
|
|
|
rtc::Optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
|
|
const SdpAudioFormat& format) {
|
|
const bool is_pcmu = STR_CASE_CMP(format.name.c_str(), "PCMU") == 0;
|
|
const bool is_pcma = STR_CASE_CMP(format.name.c_str(), "PCMA") == 0;
|
|
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 rtc::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,
|
|
rtc::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 rtc::MakeUnique<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 rtc::MakeUnique<AudioEncoderPcmA>(impl_config);
|
|
}
|
|
default: { return nullptr; }
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|