mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
6c9c958c69
This reverts commit 63d03f586b.
Reason for revert: Breaks downstream project (not backwards compatible API change)
Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
> std::map<std::string, std:string>
> with three aliases,
> cricket::CodecParameterMap
> SdpAudioFormat::Parameters
> SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}
Bug: None
Change-Id: I841735d98533d3b66850b9cfcf7ee0a99ddde078
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41377}
86 lines
3.1 KiB
C++
86 lines
3.1 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 <utility>
|
|
|
|
#include "absl/strings/match.h"
|
|
|
|
namespace webrtc {
|
|
|
|
SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
|
|
SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
|
|
|
|
SdpAudioFormat::SdpAudioFormat(absl::string_view name,
|
|
int clockrate_hz,
|
|
size_t num_channels)
|
|
: name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
|
|
|
|
SdpAudioFormat::SdpAudioFormat(absl::string_view 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(absl::string_view name,
|
|
int clockrate_hz,
|
|
size_t num_channels,
|
|
Parameters&& param)
|
|
: name(name),
|
|
clockrate_hz(clockrate_hz),
|
|
num_channels(num_channels),
|
|
parameters(std::move(param)) {}
|
|
|
|
bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
|
|
return absl::EqualsIgnoreCase(name, o.name) &&
|
|
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 absl::EqualsIgnoreCase(a.name, b.name) &&
|
|
a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
|
|
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
|