Files
platform-external-webrtc/webrtc/api/audio_codecs/audio_format.h
deadbeef cb3836773c Allow a received audio codec's payload type to change.
This will create another decoder instance, which isn't ideal, but
that's better than the current behavior where things don't work at all.

We still need to fix the root cause of the linked bug, which is that we
don't remember previous payload type mappings when generating an offer.

This CL also adds a check for what is a real error: when a payload type
that was mapped to one codec is changed to map to a different codec.

And lastly, this CL removes a DCHECK for an assumption that was not
valid: that subsequently applied codec lists will always be supersets of
previous lists.

BUG=webrtc:5847

Review-Url: https://codereview.webrtc.org/2831333002
Cr-Commit-Position: refs/heads/master@{#17897}
2017-04-26 23:28:42 +00:00

139 lines
4.9 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.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
#include <map>
#include <ostream>
#include <string>
#include <utility>
#include "webrtc/base/optional.h"
namespace webrtc {
// SDP specification for a single audio codec.
// NOTE: This class is still under development and may change without notice.
struct SdpAudioFormat {
using Parameters = std::map<std::string, std::string>;
SdpAudioFormat(const SdpAudioFormat&);
SdpAudioFormat(SdpAudioFormat&&);
SdpAudioFormat(const char* name, int clockrate_hz, size_t num_channels);
SdpAudioFormat(const std::string& name,
int clockrate_hz,
size_t num_channels);
SdpAudioFormat(const char* name,
int clockrate_hz,
size_t num_channels,
const Parameters& param);
SdpAudioFormat(const std::string& name,
int clockrate_hz,
size_t num_channels,
const Parameters& param);
~SdpAudioFormat();
// Returns true if this format is compatible with |o|. In SDP terminology:
// would it represent the same codec between an offer and an answer? As
// opposed to operator==, this method disregards codec parameters.
bool Matches(const SdpAudioFormat& o) const;
SdpAudioFormat& operator=(const SdpAudioFormat&);
SdpAudioFormat& operator=(SdpAudioFormat&&);
friend bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b);
friend bool operator!=(const SdpAudioFormat& a, const SdpAudioFormat& b) {
return !(a == b);
}
std::string name;
int clockrate_hz;
size_t num_channels;
Parameters parameters;
};
void swap(SdpAudioFormat& a, SdpAudioFormat& b);
std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf);
// Information about how an audio format is treated by the codec implementation.
// Contains basic information, such as sample rate and number of channels, which
// isn't uniformly presented by SDP. Also contains flags indicating support for
// integrating with other parts of WebRTC, like external VAD and comfort noise
// level calculation.
//
// To avoid API breakage, and make the code clearer, AudioCodecInfo should not
// be directly initializable with any flags indicating optional support. If it
// were, these initializers would break any time a new flag was added. It's also
// more difficult to understand:
// AudioCodecInfo info{16000, 1, 32000, true, false, false, true, true};
// than
// AudioCodecInfo info(16000, 1, 32000);
// info.allow_comfort_noise = true;
// info.future_flag_b = true;
// info.future_flag_c = true;
struct AudioCodecInfo {
AudioCodecInfo(int sample_rate_hz, size_t num_channels, int bitrate_bps);
AudioCodecInfo(int sample_rate_hz,
size_t num_channels,
int default_bitrate_bps,
int min_bitrate_bps,
int max_bitrate_bps);
AudioCodecInfo(const AudioCodecInfo& b) = default;
~AudioCodecInfo() = default;
bool operator==(const AudioCodecInfo& b) const {
return sample_rate_hz == b.sample_rate_hz &&
num_channels == b.num_channels &&
default_bitrate_bps == b.default_bitrate_bps &&
min_bitrate_bps == b.min_bitrate_bps &&
max_bitrate_bps == b.max_bitrate_bps &&
allow_comfort_noise == b.allow_comfort_noise &&
supports_network_adaption == b.supports_network_adaption;
}
bool operator!=(const AudioCodecInfo& b) const { return !(*this == b); }
bool HasFixedBitrate() const {
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);
return min_bitrate_bps == max_bitrate_bps;
}
int sample_rate_hz;
size_t num_channels;
int default_bitrate_bps;
int min_bitrate_bps;
int max_bitrate_bps;
bool allow_comfort_noise = true; // This codec can be used with an external
// comfort noise generator.
bool supports_network_adaption = false; // This codec can adapt to varying
// network conditions.
};
// AudioCodecSpec ties an audio format to specific information about the codec
// and its implementation.
struct AudioCodecSpec {
bool operator==(const AudioCodecSpec& b) const {
return format == b.format && info == b.info;
}
bool operator!=(const AudioCodecSpec& b) const { return !(*this == b); }
SdpAudioFormat format;
AudioCodecInfo info;
};
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_