mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
3e18c82820
This is a reland ofd7ee72041fOriginal change's description: > Reland "Remove our stream << overloads from non-test build targets." > > This is a reland ofc841d18d25> > Original change's description: > > Remove our stream << overloads from non-test build targets. > > > > Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and > > SocketAddress are kept behind gtest's #ifdef UNIT_TEST. > > > > Bug: webrtc:8982 > > Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70 > > Reviewed-on: https://webrtc-review.googlesource.com/64143 > > Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> > > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22916} > > > Bug: webrtc:8982 > Change-Id: Ibe08c6270e5e693eb661a6ce9e8f074b34ef8123 > Reviewed-on: https://webrtc-review.googlesource.com/71161 > Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> > Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22949} TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org Bug: webrtc:8982 Change-Id: I29247d1c28e99af36ef228d8c75b4adecbd7b199 Reviewed-on: https://webrtc-review.googlesource.com/72681 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23092}
97 lines
3.6 KiB
C++
97 lines
3.6 KiB
C++
/*
|
|
* 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 "api/audio_codecs/audio_format.h"
|
|
|
|
#include "common_types.h" // NOLINT(build/include)
|
|
|
|
namespace webrtc {
|
|
|
|
SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
|
|
SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
|
|
|
|
SdpAudioFormat::SdpAudioFormat(const char* name,
|
|
int clockrate_hz,
|
|
size_t num_channels)
|
|
: name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
|
|
|
|
SdpAudioFormat::SdpAudioFormat(const std::string& name,
|
|
int clockrate_hz,
|
|
size_t num_channels)
|
|
: name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
|
|
|
|
SdpAudioFormat::SdpAudioFormat(const char* name,
|
|
int clockrate_hz,
|
|
size_t num_channels,
|
|
const Parameters& param)
|
|
: name(name),
|
|
clockrate_hz(clockrate_hz),
|
|
num_channels(num_channels),
|
|
parameters(param) {}
|
|
|
|
SdpAudioFormat::SdpAudioFormat(const std::string& name,
|
|
int clockrate_hz,
|
|
size_t num_channels,
|
|
const Parameters& param)
|
|
: name(name),
|
|
clockrate_hz(clockrate_hz),
|
|
num_channels(num_channels),
|
|
parameters(param) {}
|
|
|
|
bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
|
|
return STR_CASE_CMP(name.c_str(), o.name.c_str()) == 0 &&
|
|
clockrate_hz == o.clockrate_hz && num_channels == o.num_channels;
|
|
}
|
|
|
|
SdpAudioFormat::~SdpAudioFormat() = default;
|
|
SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
|
|
SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
|
|
|
|
bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
|
|
return STR_CASE_CMP(a.name.c_str(), b.name.c_str()) == 0 &&
|
|
a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
|
|
a.parameters == b.parameters;
|
|
}
|
|
|
|
void swap(SdpAudioFormat& a, SdpAudioFormat& b) {
|
|
using std::swap;
|
|
swap(a.name, b.name);
|
|
swap(a.clockrate_hz, b.clockrate_hz);
|
|
swap(a.num_channels, b.num_channels);
|
|
swap(a.parameters, b.parameters);
|
|
}
|
|
|
|
AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
|
|
size_t num_channels,
|
|
int bitrate_bps)
|
|
: AudioCodecInfo(sample_rate_hz,
|
|
num_channels,
|
|
bitrate_bps,
|
|
bitrate_bps,
|
|
bitrate_bps) {}
|
|
|
|
AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
|
|
size_t num_channels,
|
|
int default_bitrate_bps,
|
|
int min_bitrate_bps,
|
|
int max_bitrate_bps)
|
|
: sample_rate_hz(sample_rate_hz),
|
|
num_channels(num_channels),
|
|
default_bitrate_bps(default_bitrate_bps),
|
|
min_bitrate_bps(min_bitrate_bps),
|
|
max_bitrate_bps(max_bitrate_bps) {
|
|
RTC_DCHECK_GT(sample_rate_hz, 0);
|
|
RTC_DCHECK_GT(num_channels, 0);
|
|
RTC_DCHECK_GE(min_bitrate_bps, 0);
|
|
RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
|
|
RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
|
|
}
|
|
|
|
} // namespace webrtc
|