Remove CodecInst pt.2
The following APIs on AudioCodingModule are deprecated with this CL:
static int NumberOfCodecs();
static int Codec(int, CodecInst*);
static int Codec(const char*, CodecInst*, int, size_t);
static int Codec(const char*, int, size_t);
absl::optional<CodecInst> SendCodec() const;
bool RegisterReceiveCodec(int, const SdpAudioFormat&);
int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
int UnregisterReceiveCodec(uint8_t);
int32_t ReceiveCodec(CodecInst*);
absl::optional<SdpAudioFormat> ReceiveFormat();
As well as this method on RtpRtcp module:
int32_t RegisterSendPayload(const CodecInst&);
Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}diff --git a/modules/audio_coding/acm2/acm_codec_database.cc b/modules/audio_coding/acm2/acm_codec_database.cc
deleted file mode 100644
index cada80c..0000000
--- a/modules/audio_coding/acm2/acm_codec_database.cc
+++ /dev/null
@@ -1,336 +0,0 @@
-/*
- * Copyright (c) 2012 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.
- */
-
-/*
- * This file generates databases with information about all supported audio
- * codecs.
- */
-
-// TODO(tlegrand): Change constant input pointers in all functions to constant
-// references, where appropriate.
-#include "modules/audio_coding/acm2/acm_codec_database.h"
-
-#include "absl/strings/match.h"
-#include "api/array_view.h"
-#include "modules/audio_coding/acm2/rent_a_codec.h"
-
-#if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
-#error iSAC and iSACFX codecs cannot be enabled at the same time
-#endif
-
-namespace webrtc {
-
-namespace acm2 {
-
-namespace {
-
-// Checks if the bitrate is valid for iSAC.
-bool IsISACRateValid(int rate) {
- return (rate == -1) || ((rate <= 56000) && (rate >= 10000));
-}
-
-// Checks if the bitrate is valid for iLBC.
-bool IsILBCRateValid(int rate, int frame_size_samples) {
- if (((frame_size_samples == 240) || (frame_size_samples == 480)) &&
- (rate == 13300)) {
- return true;
- } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) &&
- (rate == 15200)) {
- return true;
- } else {
- return false;
- }
-}
-
-// Checks if the bitrate is valid for Opus.
-bool IsOpusRateValid(int rate) {
- return (rate >= 6000) && (rate <= 510000);
-}
-
-} // namespace
-
-// Not yet used payload-types.
-// 83, 82, 81, 80, 79, 78, 77, 76, 75, 74, 73, 72, 71, 70, 69, 68,
-// 67, 66, 65
-
-const CodecInst ACMCodecDB::database_[] = {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- {103, "ISAC", 16000, 480, 1, 32000},
-#if (defined(WEBRTC_CODEC_ISAC))
- {104, "ISAC", 32000, 960, 1, 56000},
-#endif
-#endif
- // Mono
- {107, "L16", 8000, 80, 1, 128000},
- {108, "L16", 16000, 160, 1, 256000},
- {109, "L16", 32000, 320, 1, 512000},
- // Stereo
- {111, "L16", 8000, 80, 2, 128000},
- {112, "L16", 16000, 160, 2, 256000},
- {113, "L16", 32000, 320, 2, 512000},
- // G.711, PCM mu-law and A-law.
- // Mono
- {0, "PCMU", 8000, 160, 1, 64000},
- {8, "PCMA", 8000, 160, 1, 64000},
- // Stereo
- {110, "PCMU", 8000, 160, 2, 64000},
- {118, "PCMA", 8000, 160, 2, 64000},
-#ifdef WEBRTC_CODEC_ILBC
- {102, "ILBC", 8000, 240, 1, 13300},
-#endif
- // Mono
- {9, "G722", 16000, 320, 1, 64000},
- // Stereo
- {119, "G722", 16000, 320, 2, 64000},
-#ifdef WEBRTC_CODEC_OPUS
- // Opus internally supports 48, 24, 16, 12, 8 kHz.
- // Mono and stereo.
- {120, "opus", 48000, 960, 2, 64000},
-#endif
- // Comfort noise for four different sampling frequencies.
- {13, "CN", 8000, 240, 1, 0},
- {98, "CN", 16000, 480, 1, 0},
- {99, "CN", 32000, 960, 1, 0},
-#ifdef ENABLE_48000_HZ
- {100, "CN", 48000, 1440, 1, 0},
-#endif
- {106, "telephone-event", 8000, 240, 1, 0},
- {114, "telephone-event", 16000, 240, 1, 0},
- {115, "telephone-event", 32000, 240, 1, 0},
- {116, "telephone-event", 48000, 240, 1, 0},
-#ifdef WEBRTC_CODEC_RED
- {127, "red", 8000, 0, 1, 0},
-#endif
- // To prevent compile errors due to trailing commas.
- {-1, "Null", -1, -1, 0, -1}};
-
-// Create database with all codec settings at compile time.
-// Each entry needs the following parameters in the given order:
-// Number of allowed packet sizes, a vector with the allowed packet sizes,
-// Basic block samples, max number of channels that are supported.
-const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- {2, {480, 960}, 0, 1},
-#if (defined(WEBRTC_CODEC_ISAC))
- {1, {960}, 0, 1},
-#endif
-#endif
- // Mono
- {4, {80, 160, 240, 320}, 0, 2},
- {4, {160, 320, 480, 640}, 0, 2},
- {2, {320, 640}, 0, 2},
- // Stereo
- {4, {80, 160, 240, 320}, 0, 2},
- {4, {160, 320, 480, 640}, 0, 2},
- {2, {320, 640}, 0, 2},
- // G.711, PCM mu-law and A-law.
- // Mono
- {6, {80, 160, 240, 320, 400, 480}, 0, 2},
- {6, {80, 160, 240, 320, 400, 480}, 0, 2},
- // Stereo
- {6, {80, 160, 240, 320, 400, 480}, 0, 2},
- {6, {80, 160, 240, 320, 400, 480}, 0, 2},
-#ifdef WEBRTC_CODEC_ILBC
- {4, {160, 240, 320, 480}, 0, 1},
-#endif
- // Mono
- {6, {160, 320, 480, 640, 800, 960}, 0, 2},
- // Stereo
- {6, {160, 320, 480, 640, 800, 960}, 0, 2},
-#ifdef WEBRTC_CODEC_OPUS
-// Opus supports frames shorter than 10ms,
-// but it doesn't help us to use them.
-// Mono and stereo.
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
- {5, {480, 960, 1920, 2880, 5760}, 0, 2},
-#else
- {4, {480, 960, 1920, 2880}, 0, 2},
-#endif
-#endif
- // Comfort noise for three different sampling frequencies.
- {1, {240}, 240, 1},
- {1, {480}, 480, 1},
- {1, {960}, 960, 1},
-// TODO(solenberg): What is this flag? It is never set in the build files.
-#ifdef ENABLE_48000_HZ
- {1, {1440}, 1440, 1},
-#endif
- {1, {240}, 240, 1},
- {1, {240}, 240, 1},
- {1, {240}, 240, 1},
- {1, {240}, 240, 1},
-#ifdef WEBRTC_CODEC_RED
- {1, {0}, 0, 1},
-#endif
- // To prevent compile errors due to trailing commas.
- {-1, {-1}, -1, 0}};
-
-// Create a database of all NetEQ decoders at compile time.
-const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- NetEqDecoder::kDecoderISAC,
-#if (defined(WEBRTC_CODEC_ISAC))
- NetEqDecoder::kDecoderISACswb,
-#endif
-#endif
- // Mono
- NetEqDecoder::kDecoderPCM16B, NetEqDecoder::kDecoderPCM16Bwb,
- NetEqDecoder::kDecoderPCM16Bswb32kHz,
- // Stereo
- NetEqDecoder::kDecoderPCM16B_2ch, NetEqDecoder::kDecoderPCM16Bwb_2ch,
- NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch,
- // G.711, PCM mu-las and A-law.
- // Mono
- NetEqDecoder::kDecoderPCMu, NetEqDecoder::kDecoderPCMa,
- // Stereo
- NetEqDecoder::kDecoderPCMu_2ch, NetEqDecoder::kDecoderPCMa_2ch,
-#ifdef WEBRTC_CODEC_ILBC
- NetEqDecoder::kDecoderILBC,
-#endif
- // Mono
- NetEqDecoder::kDecoderG722,
- // Stereo
- NetEqDecoder::kDecoderG722_2ch,
-#ifdef WEBRTC_CODEC_OPUS
- // Mono and stereo.
- NetEqDecoder::kDecoderOpus,
-#endif
- // Comfort noise for three different sampling frequencies.
- NetEqDecoder::kDecoderCNGnb, NetEqDecoder::kDecoderCNGwb,
- NetEqDecoder::kDecoderCNGswb32kHz,
-#ifdef ENABLE_48000_HZ
- NetEqDecoder::kDecoderCNGswb48kHz,
-#endif
- NetEqDecoder::kDecoderAVT, NetEqDecoder::kDecoderAVT16kHz,
- NetEqDecoder::kDecoderAVT32kHz, NetEqDecoder::kDecoderAVT48kHz,
-#ifdef WEBRTC_CODEC_RED
- NetEqDecoder::kDecoderRED,
-#endif
-};
-
-// Enumerator for error codes when asking for codec database id.
-enum {
- kInvalidCodec = -10,
- kInvalidPayloadtype = -30,
- kInvalidPacketSize = -40,
- kInvalidRate = -50
-};
-
-// Gets the codec id number from the database. If there is some mismatch in
-// the codec settings, the function will return an error code.
-// NOTE! The first mismatch found will generate the return value.
-int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
- // Look for a matching codec in the database.
- int codec_id = CodecId(codec_inst);
-
- // Checks if we found a matching codec.
- if (codec_id == -1) {
- return kInvalidCodec;
- }
-
- // Checks the validity of payload type
- if (!RentACodec::IsPayloadTypeValid(codec_inst.pltype)) {
- return kInvalidPayloadtype;
- }
-
- // Comfort Noise is special case, packet-size & rate is not checked.
- if (absl::EqualsIgnoreCase(database_[codec_id].plname, "CN")) {
- return codec_id;
- }
-
- // RED is special case, packet-size & rate is not checked.
- if (absl::EqualsIgnoreCase(database_[codec_id].plname, "red")) {
- return codec_id;
- }
-
- // Checks the validity of packet size.
- if (codec_settings_[codec_id].num_packet_sizes > 0) {
- bool packet_size_ok = false;
- int i;
- int packet_size_samples;
- for (i = 0; i < codec_settings_[codec_id].num_packet_sizes; i++) {
- packet_size_samples = codec_settings_[codec_id].packet_sizes_samples[i];
- if (codec_inst.pacsize == packet_size_samples) {
- packet_size_ok = true;
- break;
- }
- }
-
- if (!packet_size_ok) {
- return kInvalidPacketSize;
- }
- }
-
- if (codec_inst.pacsize < 1) {
- return kInvalidPacketSize;
- }
-
- // Check the validity of rate. Codecs with multiple rates have their own
- // function for this.
- if (absl::EqualsIgnoreCase("isac", codec_inst.plname)) {
- return IsISACRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
- } else if (absl::EqualsIgnoreCase("ilbc", codec_inst.plname)) {
- return IsILBCRateValid(codec_inst.rate, codec_inst.pacsize) ? codec_id
- : kInvalidRate;
- } else if (absl::EqualsIgnoreCase("opus", codec_inst.plname)) {
- return IsOpusRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
- }
-
- return database_[codec_id].rate == codec_inst.rate ? codec_id : kInvalidRate;
-}
-
-// Looks for a matching payload name, frequency, and channels in the
-// codec list. Need to check all three since some codecs have several codec
-// entries with different frequencies and/or channels.
-// Does not check other codec settings, such as payload type and packet size.
-// Returns the id of the codec, or -1 if no match is found.
-int ACMCodecDB::CodecId(const CodecInst& codec_inst) {
- return (CodecId(codec_inst.plname, codec_inst.plfreq, codec_inst.channels));
-}
-
-int ACMCodecDB::CodecId(const char* payload_name,
- int frequency,
- size_t channels) {
- for (const CodecInst& ci : database_) {
- bool name_match = false;
- bool frequency_match = false;
- bool channels_match = false;
-
- // Payload name, sampling frequency and number of channels need to match.
- // NOTE! If |frequency| is -1, the frequency is not applicable, and is
- // always treated as true, like for RED.
- name_match = absl::EqualsIgnoreCase(ci.plname, payload_name);
- frequency_match = (frequency == ci.plfreq) || (frequency == -1);
- // The number of channels must match for all codecs but Opus.
- if (!absl::EqualsIgnoreCase(payload_name, "opus")) {
- channels_match = (channels == ci.channels);
- } else {
- // For opus we just check that number of channels is valid.
- channels_match = (channels == 1 || channels == 2);
- }
-
- if (name_match && frequency_match && channels_match) {
- // We have found a matching codec in the list.
- return &ci - database_;
- }
- }
-
- // We didn't find a matching codec.
- return -1;
-}
-// Gets codec id number from database for the receiver.
-int ACMCodecDB::ReceiverCodecNumber(const CodecInst& codec_inst) {
- // Look for a matching codec in the database.
- return CodecId(codec_inst);
-}
-
-} // namespace acm2
-
-} // namespace webrtc
diff --git a/modules/audio_coding/acm2/acm_codec_database.h b/modules/audio_coding/acm2/acm_codec_database.h
deleted file mode 100644
index ee6bb46..0000000
--- a/modules/audio_coding/acm2/acm_codec_database.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (c) 2012 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.
- */
-
-/*
- * This file generates databases with information about all supported audio
- * codecs.
- */
-
-#ifndef MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
-#define MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
-
-#include "common_types.h" // NOLINT(build/include)
-#include "modules/audio_coding/acm2/rent_a_codec.h"
-
-namespace webrtc {
-
-namespace acm2 {
-
-// TODO(tlegrand): replace class ACMCodecDB with a namespace.
-class ACMCodecDB {
- public:
- // kMaxNumCodecs - Maximum number of codecs that can be activated in one
- // build.
- // kMaxNumPacketSize - Maximum number of allowed packet sizes for one codec.
- // These might need to be increased if adding a new codec to the database
- static const int kMaxNumCodecs = 50;
- static const int kMaxNumPacketSize = 6;
-
- // Codec specific settings
- //
- // num_packet_sizes - number of allowed packet sizes.
- // packet_sizes_samples - list of the allowed packet sizes.
- // basic_block_samples - assigned a value different from 0 if the codec
- // requires to be fed with a specific number of samples
- // that can be different from packet size.
- // channel_support - number of channels supported to encode;
- // 1 = mono, 2 = stereo, etc.
- struct CodecSettings {
- int num_packet_sizes;
- int packet_sizes_samples[kMaxNumPacketSize];
- int basic_block_samples;
- size_t channel_support;
- };
-
- // Returns codec id from database, given the information received in the input
- // [codec_inst].
- // Input:
- // [codec_inst] - Information about the codec for which we require the
- // database id.
- // Return:
- // codec id if successful, otherwise < 0.
- static int CodecNumber(const CodecInst& codec_inst);
- static int CodecId(const CodecInst& codec_inst);
- static int CodecId(const char* payload_name, int frequency, size_t channels);
- static int ReceiverCodecNumber(const CodecInst& codec_inst);
-
- // Databases with information about the supported codecs
- // database_ - stored information about all codecs: payload type, name,
- // sampling frequency, packet size in samples, default channel
- // support, and default rate.
- // codec_settings_ - stored codec settings: number of allowed packet sizes,
- // a vector with the allowed packet sizes, basic block
- // samples, and max number of channels that are supported.
- // neteq_decoders_ - list of supported decoders in NetEQ.
- static const CodecInst database_[kMaxNumCodecs];
- static const CodecSettings codec_settings_[kMaxNumCodecs];
- static const NetEqDecoder neteq_decoders_[kMaxNumCodecs];
-};
-
-} // namespace acm2
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index c149ec1..76b0506 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -14,9 +14,7 @@
#include <memory>
-#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/packet.h"
@@ -28,89 +26,6 @@
namespace test {
namespace {
-// Returns true if the codec should be registered, otherwise false. Changes
-// the number of channels for the Opus codec to always be 1.
-bool ModifyAndUseThisCodec(CodecInst* codec_param) {
- if (absl::EqualsIgnoreCase(codec_param->plname, "CN") &&
- codec_param->plfreq == 48000)
- return false; // Skip 48 kHz comfort noise.
-
- if (absl::EqualsIgnoreCase(codec_param->plname, "telephone-event"))
- return false; // Skip DTFM.
-
- return true;
-}
-
-// Remaps payload types from ACM's default to those used in the resource file
-// neteq_universal_new.rtp. Returns true if the codec should be registered,
-// otherwise false. The payload types are set as follows (all are mono codecs):
-// PCMu = 0;
-// PCMa = 8;
-// Comfort noise 8 kHz = 13
-// Comfort noise 16 kHz = 98
-// Comfort noise 32 kHz = 99
-// iLBC = 102
-// iSAC wideband = 103
-// iSAC super-wideband = 104
-// AVT/DTMF = 106
-// RED = 117
-// PCM16b 8 kHz = 93
-// PCM16b 16 kHz = 94
-// PCM16b 32 kHz = 95
-// G.722 = 94
-bool RemapPltypeAndUseThisCodec(const char* plname,
- int plfreq,
- size_t channels,
- int* pltype) {
- if (channels != 1)
- return false; // Don't use non-mono codecs.
-
- // Re-map pltypes to those used in the NetEq test files.
- if (absl::EqualsIgnoreCase(plname, "PCMU") && plfreq == 8000) {
- *pltype = 0;
- } else if (absl::EqualsIgnoreCase(plname, "PCMA") && plfreq == 8000) {
- *pltype = 8;
- } else if (absl::EqualsIgnoreCase(plname, "CN") && plfreq == 8000) {
- *pltype = 13;
- } else if (absl::EqualsIgnoreCase(plname, "CN") && plfreq == 16000) {
- *pltype = 98;
- } else if (absl::EqualsIgnoreCase(plname, "CN") && plfreq == 32000) {
- *pltype = 99;
- } else if (absl::EqualsIgnoreCase(plname, "ILBC")) {
- *pltype = 102;
- } else if (absl::EqualsIgnoreCase(plname, "ISAC") && plfreq == 16000) {
- *pltype = 103;
- } else if (absl::EqualsIgnoreCase(plname, "ISAC") && plfreq == 32000) {
- *pltype = 104;
- } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
- plfreq == 8000) {
- *pltype = 106;
- } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
- plfreq == 16000) {
- *pltype = 114;
- } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
- plfreq == 32000) {
- *pltype = 115;
- } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
- plfreq == 48000) {
- *pltype = 116;
- } else if (absl::EqualsIgnoreCase(plname, "red")) {
- *pltype = 117;
- } else if (absl::EqualsIgnoreCase(plname, "L16") && plfreq == 8000) {
- *pltype = 93;
- } else if (absl::EqualsIgnoreCase(plname, "L16") && plfreq == 16000) {
- *pltype = 94;
- } else if (absl::EqualsIgnoreCase(plname, "L16") && plfreq == 32000) {
- *pltype = 95;
- } else if (absl::EqualsIgnoreCase(plname, "G722")) {
- *pltype = 9;
- } else {
- // Don't use any other codecs.
- return false;
- }
- return true;
-}
-
AudioCodingModule::Config MakeAcmConfig(
Clock* clock,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
@@ -119,7 +34,6 @@
config.decoder_factory = std::move(decoder_factory);
return config;
}
-
} // namespace
AcmReceiveTestOldApi::AcmReceiveTestOldApi(
@@ -139,36 +53,43 @@
AcmReceiveTestOldApi::~AcmReceiveTestOldApi() = default;
void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
- CodecInst my_codec_param;
- for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
- ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
- if (ModifyAndUseThisCodec(&my_codec_param)) {
- ASSERT_EQ(true,
- acm_->RegisterReceiveCodec(my_codec_param.pltype,
- CodecInstToSdp(my_codec_param)))
- << "Couldn't register receive codec.\n";
- }
- }
+ acm_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
+ {104, {"ISAC", 32000, 1}},
+ {107, {"L16", 8000, 1}},
+ {108, {"L16", 16000, 1}},
+ {109, {"L16", 32000, 1}},
+ {111, {"L16", 8000, 2}},
+ {112, {"L16", 16000, 2}},
+ {113, {"L16", 32000, 2}},
+ {0, {"PCMU", 8000, 1}},
+ {110, {"PCMU", 8000, 2}},
+ {8, {"PCMA", 8000, 1}},
+ {118, {"PCMA", 8000, 2}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {119, {"G722", 8000, 2}},
+ {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
}
+// Remaps payload types from ACM's default to those used in the resource file
+// neteq_universal_new.rtp.
void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
- CodecInst my_codec_param;
- for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
- ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
- if (!ModifyAndUseThisCodec(&my_codec_param)) {
- // Skip this codec.
- continue;
- }
-
- if (RemapPltypeAndUseThisCodec(my_codec_param.plname, my_codec_param.plfreq,
- my_codec_param.channels,
- &my_codec_param.pltype)) {
- ASSERT_EQ(true,
- acm_->RegisterReceiveCodec(my_codec_param.pltype,
- CodecInstToSdp(my_codec_param)))
- << "Couldn't register receive codec.\n";
- }
- }
+ acm_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
+ {104, {"ISAC", 32000, 1}},
+ {93, {"L16", 8000, 1}},
+ {94, {"L16", 16000, 1}},
+ {95, {"L16", 32000, 1}},
+ {0, {"PCMU", 8000, 1}},
+ {8, {"PCMA", 8000, 1}},
+ {102, {"ILBC", 8000, 1}},
+ {9, {"G722", 8000, 1}},
+ {120, {"OPUS", 48000, 2}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
}
void AcmReceiveTestOldApi::Run() {
diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h
index 83ffcb3..04aefc7 100644
--- a/modules/audio_coding/acm2/acm_receive_test.h
+++ b/modules/audio_coding/acm2/acm_receive_test.h
@@ -23,7 +23,6 @@
namespace webrtc {
class AudioCodingModule;
class AudioDecoder;
-struct CodecInst;
namespace test {
class AudioSink;
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index d3af7c0..ea84036 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -20,7 +20,6 @@
#include "api/audio_codecs/audio_decoder.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
-#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
#include "modules/include/module_common_types.h"
@@ -62,7 +61,10 @@
absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
rtc::CritScope lock(&crit_sect_);
- return last_packet_sample_rate_hz_;
+ if (!last_decoder_) {
+ return absl::nullopt;
+ }
+ return last_decoder_->second.clockrate_hz;
}
int AcmReceiver::last_output_sample_rate_hz() const {
@@ -71,45 +73,44 @@
int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> incoming_payload) {
- uint32_t receive_timestamp = 0;
- const RTPHeader* header = &rtp_header.header; // Just a shorthand.
-
if (incoming_payload.empty()) {
neteq_->InsertEmptyPacket(rtp_header.header);
return 0;
}
+ const RTPHeader& header = rtp_header.header; // Just a shorthand.
+ int payload_type = header.payloadType;
+ auto format = neteq_->GetDecoderFormat(payload_type);
+ if (format && absl::EqualsIgnoreCase(format->name, "red")) {
+ // This is a RED packet. Get the format of the audio codec.
+ payload_type = incoming_payload[0] & 0x7f;
+ format = neteq_->GetDecoderFormat(payload_type);
+ }
+ if (!format) {
+ RTC_LOG_F(LS_ERROR) << "Payload-type "
+ << payload_type
+ << " is not registered.";
+ return -1;
+ }
+
{
rtc::CritScope lock(&crit_sect_);
-
- const absl::optional<CodecInst> ci =
- RtpHeaderToDecoder(*header, incoming_payload[0]);
- if (!ci) {
- RTC_LOG_F(LS_ERROR) << "Payload-type "
- << static_cast<int>(header->payloadType)
- << " is not registered.";
- return -1;
- }
- receive_timestamp = NowInTimestamp(ci->plfreq);
-
- if (absl::EqualsIgnoreCase(ci->plname, "cn")) {
- if (last_audio_decoder_ && last_audio_decoder_->channels > 1) {
+ if (absl::EqualsIgnoreCase(format->name, "cn")) {
+ if (last_decoder_ && last_decoder_->second.num_channels > 1) {
// This is a CNG and the audio codec is not mono, so skip pushing in
// packets into NetEq.
return 0;
}
} else {
- last_audio_decoder_ = ci;
- last_audio_format_ = neteq_->GetDecoderFormat(ci->pltype);
- RTC_DCHECK(last_audio_format_);
- last_packet_sample_rate_hz_ = ci->plfreq;
+ RTC_DCHECK(format);
+ last_decoder_ = std::make_pair(payload_type, *format);
}
} // |crit_sect_| is released.
- if (neteq_->InsertPacket(rtp_header.header, incoming_payload,
- receive_timestamp) < 0) {
+ uint32_t receive_timestamp = NowInTimestamp(format->clockrate_hz);
+ if (neteq_->InsertPacket(header, incoming_payload, receive_timestamp) < 0) {
RTC_LOG(LERROR) << "AcmReceiver::InsertPacket "
- << static_cast<int>(header->payloadType)
+ << static_cast<int>(header.payloadType)
<< " Failed to insert packet";
return -1;
}
@@ -186,85 +187,6 @@
neteq_->SetCodecs(codecs);
}
-int32_t AcmReceiver::AddCodec(int acm_codec_id,
- uint8_t payload_type,
- size_t channels,
- int /*sample_rate_hz*/,
- AudioDecoder* audio_decoder,
- const std::string& name) {
- // TODO(kwiberg): This function has been ignoring the |sample_rate_hz|
- // argument for a long time. Arguably, it should simply be removed.
-
- const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
- if (acm_codec_id == -1)
- return NetEqDecoder::kDecoderArbitrary; // External decoder.
- const absl::optional<RentACodec::CodecId> cid =
- RentACodec::CodecIdFromIndex(acm_codec_id);
- RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id;
- const absl::optional<NetEqDecoder> ned =
- RentACodec::NetEqDecoderFromCodecId(*cid, channels);
- RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid);
- return *ned;
- }();
- const absl::optional<SdpAudioFormat> new_format =
- NetEqDecoderToSdpAudioFormat(neteq_decoder);
-
- rtc::CritScope lock(&crit_sect_);
-
- const auto old_format = neteq_->GetDecoderFormat(payload_type);
- if (old_format && new_format && *old_format == *new_format) {
- // Re-registering the same codec. Do nothing and return.
- return 0;
- }
-
- if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
- RTC_LOG(LERROR) << "Cannot remove payload "
- << static_cast<int>(payload_type);
- return -1;
- }
-
- int ret_val;
- if (!audio_decoder) {
- ret_val = neteq_->RegisterPayloadType(neteq_decoder, name, payload_type);
- } else {
- ret_val = neteq_->RegisterExternalDecoder(audio_decoder, neteq_decoder,
- name, payload_type);
- }
- if (ret_val != NetEq::kOK) {
- RTC_LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
- << static_cast<int>(payload_type)
- << " channels: " << channels;
- return -1;
- }
- return 0;
-}
-
-bool AcmReceiver::AddCodec(int rtp_payload_type,
- const SdpAudioFormat& audio_format) {
- const auto old_format = neteq_->GetDecoderFormat(rtp_payload_type);
- if (old_format && *old_format == audio_format) {
- // Re-registering the same codec. Do nothing and return.
- return true;
- }
-
- if (neteq_->RemovePayloadType(rtp_payload_type) != NetEq::kOK) {
- RTC_LOG(LERROR)
- << "AcmReceiver::AddCodec: Could not remove existing decoder"
- " for payload type "
- << rtp_payload_type;
- return false;
- }
-
- const bool success =
- neteq_->RegisterPayloadType(rtp_payload_type, audio_format);
- if (!success) {
- RTC_LOG(LERROR) << "AcmReceiver::AddCodec failed for payload type "
- << rtp_payload_type << ", decoder format "
- << rtc::ToString(audio_format);
- }
- return success;
-}
-
void AcmReceiver::FlushBuffers() {
neteq_->FlushBuffers();
}
@@ -272,24 +194,7 @@
void AcmReceiver::RemoveAllCodecs() {
rtc::CritScope lock(&crit_sect_);
neteq_->RemoveAllPayloadTypes();
- last_audio_decoder_ = absl::nullopt;
- last_audio_format_ = absl::nullopt;
- last_packet_sample_rate_hz_ = absl::nullopt;
-}
-
-int AcmReceiver::RemoveCodec(uint8_t payload_type) {
- rtc::CritScope lock(&crit_sect_);
- if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
- RTC_LOG(LERROR) << "AcmReceiver::RemoveCodec "
- << static_cast<int>(payload_type);
- return -1;
- }
- if (last_audio_decoder_ && payload_type == last_audio_decoder_->pltype) {
- last_audio_decoder_ = absl::nullopt;
- last_audio_format_ = absl::nullopt;
- last_packet_sample_rate_hz_ = absl::nullopt;
- }
- return 0;
+ last_decoder_ = absl::nullopt;
}
absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
@@ -304,18 +209,14 @@
return neteq_->TargetDelayMs();
}
-int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
+absl::optional<std::pair<int, SdpAudioFormat>>
+ AcmReceiver::LastDecoder() const {
rtc::CritScope lock(&crit_sect_);
- if (!last_audio_decoder_) {
- return -1;
+ if (!last_decoder_) {
+ return absl::nullopt;
}
- *codec = *last_audio_decoder_;
- return 0;
-}
-
-absl::optional<SdpAudioFormat> AcmReceiver::LastAudioFormat() const {
- rtc::CritScope lock(&crit_sect_);
- return last_audio_format_;
+ RTC_DCHECK_NE(-1, last_decoder_->first); // Payload type should be valid.
+ return last_decoder_;
}
void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
@@ -354,26 +255,6 @@
neteq_operations_and_state.packet_buffer_flushes;
}
-int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
- CodecInst* codec) const {
- rtc::CritScope lock(&crit_sect_);
- const absl::optional<CodecInst> ci = neteq_->GetDecoder(payload_type);
- if (ci) {
- *codec = *ci;
- return 0;
- } else {
- RTC_LOG(LERROR) << "AcmReceiver::DecoderByPayloadType "
- << static_cast<int>(payload_type);
- return -1;
- }
-}
-
-absl::optional<SdpAudioFormat> AcmReceiver::DecoderByPayloadType(
- int payload_type) const {
- rtc::CritScope lock(&crit_sect_);
- return neteq_->GetDecoderFormat(payload_type);
-}
-
int AcmReceiver::EnableNack(size_t max_nack_list_size) {
neteq_->EnableNack(max_nack_list_size);
return 0;
@@ -393,19 +274,6 @@
// TODO(turajs): Should NetEq Buffer be flushed?
}
-const absl::optional<CodecInst> AcmReceiver::RtpHeaderToDecoder(
- const RTPHeader& rtp_header,
- uint8_t first_payload_byte) const {
- const absl::optional<CodecInst> ci =
- neteq_->GetDecoder(rtp_header.payloadType);
- if (ci && absl::EqualsIgnoreCase(ci->plname, "red")) {
- // This is a RED packet. Get the payload of the audio codec.
- return neteq_->GetDecoder(first_payload_byte & 0x7f);
- } else {
- return ci;
- }
-}
-
uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
// Down-cast the time to (32-6)-bit since we only care about
// the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 8e7d839..0cbaa9a 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -15,6 +15,7 @@
#include <map>
#include <memory>
#include <string>
+#include <utility>
#include <vector>
#include "absl/types/optional.h"
@@ -30,7 +31,6 @@
namespace webrtc {
class Clock;
-struct CodecInst;
class NetEq;
struct RTPHeader;
struct WebRtcRTPHeader;
@@ -85,43 +85,6 @@
void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
//
- // Adds a new codec to the NetEq codec database.
- //
- // Input:
- // - acm_codec_id : ACM codec ID; -1 means external decoder.
- // - payload_type : payload type.
- // - sample_rate_hz : sample rate.
- // - audio_decoder : pointer to a decoder object. If it's null, then
- // NetEq will internally create a decoder object
- // based on the value of |acm_codec_id| (which
- // mustn't be -1). Otherwise, NetEq will use the
- // given decoder for the given payload type. NetEq
- // won't take ownership of the decoder; it's up to
- // the caller to delete it when it's no longer
- // needed.
- //
- // Providing an existing decoder object here is
- // necessary for external decoders, but may also be
- // used for built-in decoders if NetEq doesn't have
- // all the info it needs to construct them properly
- // (e.g. iSAC, where the decoder needs to be paired
- // with an encoder).
- //
- // Return value : 0 if OK.
- // <0 if NetEq returned an error.
- //
- int AddCodec(int acm_codec_id,
- uint8_t payload_type,
- size_t channels,
- int sample_rate_hz,
- AudioDecoder* audio_decoder,
- const std::string& name);
-
- // Adds a new decoder to the NetEq codec database. Returns true iff
- // successful.
- bool AddCodec(int rtp_payload_type, const SdpAudioFormat& audio_format);
-
- //
// Sets a minimum delay for packet buffer. The given delay is maintained,
// unless channel condition dictates a higher delay.
//
@@ -173,17 +136,6 @@
void FlushBuffers();
//
- // Removes a payload-type from the NetEq codec database.
- //
- // Input:
- // - payload_type : the payload-type to be removed.
- //
- // Return value : 0 if OK.
- // -1 if an error occurred.
- //
- int RemoveCodec(uint8_t payload_type);
-
- //
// Remove all registered codecs.
//
void RemoveAllCodecs();
@@ -204,30 +156,10 @@
int TargetDelayMs() const;
//
- // Get the audio codec associated with the last non-CNG/non-DTMF received
- // payload. If no non-CNG/non-DTMF packet is received -1 is returned,
- // otherwise return 0.
+ // Get payload type and format of the last non-CNG/non-DTMF received payload.
+ // If no non-CNG/non-DTMF packet is received absl::nullopt is returned.
//
- int LastAudioCodec(CodecInst* codec) const;
-
- absl::optional<SdpAudioFormat> LastAudioFormat() const;
-
- //
- // Get a decoder given its registered payload-type.
- //
- // Input:
- // -payload_type : the payload-type of the codec to be retrieved.
- //
- // Output:
- // -codec : codec associated with the given payload-type.
- //
- // Return value : 0 if succeeded.
- // -1 if failed, e.g. given payload-type is not
- // registered.
- //
- int DecoderByPayloadType(uint8_t payload_type,
- CodecInst* codec) const;
- absl::optional<SdpAudioFormat> DecoderByPayloadType(int payload_type) const;
+ absl::optional<std::pair<int, SdpAudioFormat>> LastDecoder() const;
//
// Enable NACK and set the maximum size of the NACK list. If NACK is already
@@ -260,32 +192,17 @@
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
private:
- struct Decoder {
- int acm_codec_id;
- uint8_t payload_type;
- // This field is meaningful for codecs where both mono and
- // stereo versions are registered under the same ID.
- size_t channels;
- int sample_rate_hz;
- };
-
- const absl::optional<CodecInst> RtpHeaderToDecoder(
- const RTPHeader& rtp_header,
- uint8_t first_payload_byte) const
- RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
-
uint32_t NowInTimestamp(int decoder_sampling_rate) const;
rtc::CriticalSection crit_sect_;
- absl::optional<CodecInst> last_audio_decoder_ RTC_GUARDED_BY(crit_sect_);
- absl::optional<SdpAudioFormat> last_audio_format_ RTC_GUARDED_BY(crit_sect_);
+ absl::optional<std::pair<int, SdpAudioFormat>> last_decoder_
+ RTC_GUARDED_BY(crit_sect_);
ACMResampler resampler_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(crit_sect_);
CallStatistics call_stats_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<NetEq> neteq_; // NetEq is thread-safe; no lock needed.
const Clock* const clock_;
bool resampled_last_output_frame_ RTC_GUARDED_BY(crit_sect_);
- absl::optional<int> last_packet_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
};
} // namespace acm2
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 46384fe..8b3d2e4 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -15,7 +15,6 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
@@ -105,14 +104,6 @@
return num_10ms_frames;
}
- template <size_t N>
- void AddSetOfCodecs(rtc::ArrayView<SdpAudioFormat> formats) {
- static int payload_type = 0;
- for (const auto& format : formats) {
- EXPECT_TRUE(receiver_->AddCodec(payload_type++, format));
- }
- }
-
int SendData(FrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
@@ -154,111 +145,26 @@
};
#if defined(WEBRTC_ANDROID)
-#define MAYBE_AddCodecGetCodec DISABLED_AddCodecGetCodec
-#else
-#define MAYBE_AddCodecGetCodec AddCodecGetCodec
-#endif
-TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
- const std::vector<AudioCodecSpec> codecs =
- decoder_factory_->GetSupportedDecoders();
-
- // Add codec.
- for (size_t n = 0; n < codecs.size(); ++n) {
- if (n & 0x1) { // Just add codecs with odd index.
- const int payload_type = rtc::checked_cast<int>(n);
- EXPECT_TRUE(receiver_->AddCodec(payload_type, codecs[n].format));
- }
- }
- // Get codec and compare.
- for (size_t n = 0; n < codecs.size(); ++n) {
- const int payload_type = rtc::checked_cast<int>(n);
- if (n & 0x1) {
- // Codecs with odd index should match the reference.
- EXPECT_EQ(absl::make_optional(codecs[n].format),
- receiver_->DecoderByPayloadType(payload_type));
- } else {
- // Codecs with even index are not registered.
- EXPECT_EQ(absl::nullopt, receiver_->DecoderByPayloadType(payload_type));
- }
- }
-}
-
-#if defined(WEBRTC_ANDROID)
-#define MAYBE_AddCodecChangePayloadType DISABLED_AddCodecChangePayloadType
-#else
-#define MAYBE_AddCodecChangePayloadType AddCodecChangePayloadType
-#endif
-TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecChangePayloadType) {
- const SdpAudioFormat format("giraffe", 8000, 1);
-
- // Register the same codec with different payloads.
- EXPECT_EQ(true, receiver_->AddCodec(17, format));
- EXPECT_EQ(true, receiver_->AddCodec(18, format));
-
- // Both payload types should exist.
- EXPECT_EQ(absl::make_optional(format), receiver_->DecoderByPayloadType(17));
- EXPECT_EQ(absl::make_optional(format), receiver_->DecoderByPayloadType(18));
-}
-
-#if defined(WEBRTC_ANDROID)
-#define MAYBE_AddCodecChangeCodecId DISABLED_AddCodecChangeCodecId
-#else
-#define MAYBE_AddCodecChangeCodecId AddCodecChangeCodecId
-#endif
-TEST_F(AcmReceiverTestOldApi, AddCodecChangeCodecId) {
- const SdpAudioFormat format1("giraffe", 8000, 1);
- const SdpAudioFormat format2("gnu", 16000, 1);
-
- // Register the same payload type with different codec ID.
- EXPECT_EQ(true, receiver_->AddCodec(17, format1));
- EXPECT_EQ(true, receiver_->AddCodec(17, format2));
-
- // Make sure that the last codec is used.
- EXPECT_EQ(absl::make_optional(format2), receiver_->DecoderByPayloadType(17));
-}
-
-#if defined(WEBRTC_ANDROID)
-#define MAYBE_AddCodecRemoveCodec DISABLED_AddCodecRemoveCodec
-#else
-#define MAYBE_AddCodecRemoveCodec AddCodecRemoveCodec
-#endif
-TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecRemoveCodec) {
- EXPECT_EQ(true, receiver_->AddCodec(17, SdpAudioFormat("giraffe", 8000, 1)));
-
- // Remove non-existing codec should not fail. ACM1 legacy.
- EXPECT_EQ(0, receiver_->RemoveCodec(18));
-
- // Remove an existing codec.
- EXPECT_EQ(0, receiver_->RemoveCodec(17));
-
- // Ask for the removed codec, must fail.
- EXPECT_EQ(absl::nullopt, receiver_->DecoderByPayloadType(17));
-}
-
-#if defined(WEBRTC_ANDROID)
#define MAYBE_SampleRate DISABLED_SampleRate
#else
#define MAYBE_SampleRate SampleRate
#endif
TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
- const std::vector<SdpAudioFormat> codecs = {{"ISAC", 16000, 1},
- {"ISAC", 32000, 1}};
- for (size_t i = 0; i < codecs.size(); ++i) {
- const int payload_type = rtc::checked_cast<int>(i);
- EXPECT_EQ(true, receiver_->AddCodec(payload_type, codecs[i]));
- }
+ const std::map<int, SdpAudioFormat> codecs = {{0, {"ISAC", 16000, 1}},
+ {1, {"ISAC", 32000, 1}}};
+ receiver_->SetCodecs(codecs);
constexpr int kOutSampleRateHz = 8000; // Different than codec sample rate.
for (size_t i = 0; i < codecs.size(); ++i) {
const int payload_type = rtc::checked_cast<int>(i);
const int num_10ms_frames =
- InsertOnePacketOfSilence(SetEncoder(payload_type, codecs[i]));
+ InsertOnePacketOfSilence(SetEncoder(payload_type, codecs.at(i)));
for (int k = 0; k < num_10ms_frames; ++k) {
AudioFrame frame;
bool muted;
EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame, &muted));
}
- EXPECT_EQ(encoder_factory_->QueryAudioEncoder(codecs[i])->sample_rate_hz,
+ EXPECT_EQ(encoder_factory_->QueryAudioEncoder(codecs.at(i))->sample_rate_hz,
receiver_->last_output_sample_rate_hz());
}
}
@@ -278,7 +184,7 @@
EXPECT_TRUE(config_.neteq_config.for_test_no_time_stretching);
constexpr int payload_type = 17;
- EXPECT_TRUE(receiver_->AddCodec(payload_type, codec));
+ receiver_->SetCodecs({{payload_type, codec}});
const AudioCodecInfo info = SetEncoder(payload_type, codec);
const int output_sample_rate_hz = info.sample_rate_hz;
@@ -356,7 +262,7 @@
constexpr int payload_type = 34;
const SdpAudioFormat codec = {"L16", 16000, 1};
const AudioCodecInfo info = SetEncoder(payload_type, codec);
- EXPECT_TRUE(receiver_->AddCodec(payload_type, codec));
+ receiver_->SetCodecs({{payload_type, codec}});
constexpr int kNumPackets = 5;
AudioFrame frame;
for (int n = 0; n < kNumPackets; ++n) {
@@ -387,7 +293,7 @@
const SdpAudioFormat codec = {"L16", 16000, 1};
const AudioCodecInfo info = SetEncoder(payload_type, codec);
encoder_factory_->QueryAudioEncoder(codec).value();
- EXPECT_TRUE(receiver_->AddCodec(payload_type, codec));
+ receiver_->SetCodecs({{payload_type, codec}});
const int kNumPackets = 5;
AudioFrame frame;
for (int n = 0; n < kNumPackets; ++n) {
@@ -407,43 +313,43 @@
#endif
#if defined(WEBRTC_CODEC_ISAC)
TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
- const std::vector<SdpAudioFormat> codecs = {{"ISAC", 16000, 1},
- {"PCMA", 8000, 1},
- {"ISAC", 32000, 1},
- {"L16", 32000, 1}};
- for (size_t i = 0; i < codecs.size(); ++i) {
- const int payload_type = rtc::checked_cast<int>(i);
- EXPECT_TRUE(receiver_->AddCodec(payload_type, codecs[i]));
- }
-
- const std::map<int, int> cng_payload_types = {
- {8000, 100}, {16000, 101}, {32000, 102}};
- for (const auto& x : cng_payload_types) {
- const int sample_rate_hz = x.first;
- const int payload_type = x.second;
- EXPECT_TRUE(receiver_->AddCodec(payload_type, {"CN", sample_rate_hz, 1}));
+ const std::map<int, SdpAudioFormat> codecs = {{0, {"ISAC", 16000, 1}},
+ {1, {"PCMA", 8000, 1}},
+ {2, {"ISAC", 32000, 1}},
+ {3, {"L16", 32000, 1}}};
+ const std::map<int, int> cng_payload_types = {{8000, 100},
+ {16000, 101},
+ {32000, 102}};
+ {
+ std::map<int, SdpAudioFormat> receive_codecs = codecs;
+ for (const auto& cng_type : cng_payload_types) {
+ receive_codecs.emplace(
+ std::make_pair(cng_type.second, SdpAudioFormat("CN", cng_type.first, 1)));
+ }
+ receiver_->SetCodecs(receive_codecs);
}
// No audio payload is received.
- EXPECT_EQ(absl::nullopt, receiver_->LastAudioFormat());
+ EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
// Start with sending DTX.
packet_sent_ = false;
InsertOnePacketOfSilence(
- SetEncoder(0, codecs[0], cng_payload_types)); // Enough to test
+ SetEncoder(0, codecs.at(0), cng_payload_types)); // Enough to test
// with one codec.
ASSERT_TRUE(packet_sent_);
EXPECT_EQ(kAudioFrameCN, last_frame_type_);
// Has received, only, DTX. Last Audio codec is undefined.
- EXPECT_EQ(absl::nullopt, receiver_->LastAudioFormat());
- EXPECT_FALSE(receiver_->last_packet_sample_rate_hz());
+ EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
+ EXPECT_EQ(absl::nullopt, receiver_->last_packet_sample_rate_hz());
for (size_t i = 0; i < codecs.size(); ++i) {
// Set DTX off to send audio payload.
packet_sent_ = false;
const int payload_type = rtc::checked_cast<int>(i);
- const AudioCodecInfo info_without_cng = SetEncoder(payload_type, codecs[i]);
+ const AudioCodecInfo info_without_cng =
+ SetEncoder(payload_type, codecs.at(i));
InsertOnePacketOfSilence(info_without_cng);
// Sanity check if Actually an audio payload received, and it should be
@@ -456,7 +362,7 @@
// Set VAD on to send DTX. Then check if the "Last Audio codec" returns
// the expected codec. Encode repeatedly until a DTX is sent.
const AudioCodecInfo info_with_cng =
- SetEncoder(payload_type, codecs[i], cng_payload_types);
+ SetEncoder(payload_type, codecs.at(i), cng_payload_types);
while (last_frame_type_ != kAudioFrameCN) {
packet_sent_ = false;
InsertOnePacketOfSilence(info_with_cng);
@@ -464,7 +370,7 @@
}
EXPECT_EQ(info_with_cng.sample_rate_hz,
receiver_->last_packet_sample_rate_hz());
- EXPECT_EQ(codecs[i], receiver_->LastAudioFormat());
+ EXPECT_EQ(codecs.at(i), receiver_->LastDecoder()->second);
}
}
#endif
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index c5e010c..cf9c8eb 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -14,10 +14,10 @@
#include <stdio.h>
#include <string.h>
+#include "absl/strings/match.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/packet.h"
@@ -59,23 +59,28 @@
AcmSendTestOldApi::~AcmSendTestOldApi() = default;
bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
- int sampling_freq_hz,
- int channels,
+ int clockrate_hz,
+ int num_channels,
int payload_type,
int frame_size_samples) {
- CodecInst codec;
- RTC_CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec,
- sampling_freq_hz, channels));
- codec.pltype = payload_type;
- codec.pacsize = frame_size_samples;
- auto factory = CreateBuiltinAudioEncoderFactory();
- SdpAudioFormat format = CodecInstToSdp(codec);
+ SdpAudioFormat format(payload_name, clockrate_hz, num_channels);
+ if (absl::EqualsIgnoreCase(payload_name, "g722")) {
+ RTC_CHECK_EQ(16000, clockrate_hz);
+ format.clockrate_hz = 8000;
+ } else if (absl::EqualsIgnoreCase(payload_name, "opus")) {
+ RTC_CHECK(num_channels == 1 || num_channels == 2);
+ if (num_channels == 2) {
+ format.parameters["stereo"] = "1";
+ }
+ format.num_channels = 2;
+ }
format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
- frame_size_samples, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
+ frame_size_samples, rtc::CheckedDivExact(clockrate_hz, 1000)));
+ auto factory = CreateBuiltinAudioEncoderFactory();
acm_->SetEncoder(
factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
codec_registered_ = true;
- input_frame_.num_channels_ = channels;
+ input_frame_.num_channels_ = num_channels;
assert(input_block_size_samples_ * input_frame_.num_channels_ <=
AudioFrame::kMaxDataSizeSamples);
return codec_registered_;
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index c0aab3a..60da679 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -18,7 +18,6 @@
#include "api/array_view.h"
#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
-#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "modules/include/module_common_types.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/buffer.h"
@@ -45,9 +44,6 @@
void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
modifier) override;
- // Get current send codec.
- absl::optional<CodecInst> SendCodec() const override;
-
// Sets the bitrate to the specified value in bits/sec. In case the codec does
// not support the requested value it will choose an appropriate value
// instead.
@@ -90,19 +86,8 @@
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
- bool RegisterReceiveCodec(int rtp_payload_type,
- const SdpAudioFormat& audio_format) override;
-
- int RegisterExternalReceiveCodec(int rtp_payload_type,
- AudioDecoder* external_decoder,
- int sample_rate_hz,
- int num_channels,
- const std::string& name) override;
-
// Get current received codec.
- int ReceiveCodec(CodecInst* current_codec) const override;
-
- absl::optional<SdpAudioFormat> ReceiveFormat() const override;
+ absl::optional<std::pair<int, SdpAudioFormat>> ReceiveCodec() const override;
// Incoming packet from network parsed and ready for decode.
int IncomingPacket(const uint8_t* incoming_payload,
@@ -141,8 +126,6 @@
int DisableOpusDtx() override;
- int UnregisterReceiveCodec(uint8_t payload_type) override;
-
int EnableNack(size_t max_nack_list_size) override;
void DisableNack() override;
@@ -317,42 +300,6 @@
}
}
-// Wraps a raw AudioEncoder pointer. The idea is that you can put one of these
-// in a unique_ptr, to protect the contained raw pointer from being deleted
-// when the unique_ptr expires. (This is of course a bad idea in general, but
-// backwards compatibility.)
-class RawAudioEncoderWrapper final : public AudioEncoder {
- public:
- RawAudioEncoderWrapper(AudioEncoder* enc) : enc_(enc) {}
- int SampleRateHz() const override { return enc_->SampleRateHz(); }
- size_t NumChannels() const override { return enc_->NumChannels(); }
- int RtpTimestampRateHz() const override { return enc_->RtpTimestampRateHz(); }
- size_t Num10MsFramesInNextPacket() const override {
- return enc_->Num10MsFramesInNextPacket();
- }
- size_t Max10MsFramesInAPacket() const override {
- return enc_->Max10MsFramesInAPacket();
- }
- int GetTargetBitrate() const override { return enc_->GetTargetBitrate(); }
- EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
- rtc::ArrayView<const int16_t> audio,
- rtc::Buffer* encoded) override {
- return enc_->Encode(rtp_timestamp, audio, encoded);
- }
- void Reset() override { return enc_->Reset(); }
- bool SetFec(bool enable) override { return enc_->SetFec(enable); }
- bool SetDtx(bool enable) override { return enc_->SetDtx(enable); }
- bool SetApplication(Application application) override {
- return enc_->SetApplication(application);
- }
- void SetMaxPlaybackRate(int frequency_hz) override {
- return enc_->SetMaxPlaybackRate(frequency_hz);
- }
-
- private:
- AudioEncoder* enc_;
-};
-
void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
if (value != last_value_ || first_time_) {
first_time_ = false;
@@ -480,26 +427,6 @@
modifier(&encoder_stack_);
}
-// Get current send codec.
-absl::optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
- rtc::CritScope lock(&acm_crit_sect_);
- if (encoder_stack_) {
- CodecInst ci;
- ci.channels = encoder_stack_->NumChannels();
- ci.plfreq = encoder_stack_->SampleRateHz();
- ci.pacsize = rtc::CheckedDivExact(
- static_cast<int>(encoder_stack_->Max10MsFramesInAPacket() * ci.plfreq),
- 100);
- ci.pltype = -1; // Not valid.
- ci.rate = -1; // Not valid.
- static const char kName[] = "external";
- memcpy(ci.plname, kName, sizeof(kName));
- return ci;
- } else {
- return absl::nullopt;
- }
-}
-
void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
rtc::CritScope lock(&acm_crit_sect_);
if (encoder_stack_) {
@@ -748,54 +675,10 @@
receiver_.SetCodecs(codecs);
}
-bool AudioCodingModuleImpl::RegisterReceiveCodec(
- int rtp_payload_type,
- const SdpAudioFormat& audio_format) {
+absl::optional<std::pair<int, SdpAudioFormat>>
+ AudioCodingModuleImpl::ReceiveCodec() const {
rtc::CritScope lock(&acm_crit_sect_);
- RTC_DCHECK(receiver_initialized_);
-
- if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
- RTC_LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
- << " for decoder.";
- return false;
- }
-
- return receiver_.AddCodec(rtp_payload_type, audio_format);
-}
-
-int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
- int rtp_payload_type,
- AudioDecoder* external_decoder,
- int sample_rate_hz,
- int num_channels,
- const std::string& name) {
- rtc::CritScope lock(&acm_crit_sect_);
- RTC_DCHECK(receiver_initialized_);
- if (num_channels > 2 || num_channels < 0) {
- RTC_LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
- return -1;
- }
-
- // Check if the payload-type is valid.
- if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
- RTC_LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
- << " for external decoder.";
- return -1;
- }
-
- return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels,
- sample_rate_hz, external_decoder, name);
-}
-
-// Get current received codec.
-int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
- rtc::CritScope lock(&acm_crit_sect_);
- return receiver_.LastAudioCodec(current_codec);
-}
-
-absl::optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
- rtc::CritScope lock(&acm_crit_sect_);
- return receiver_.LastAudioFormat();
+ return receiver_.LastDecoder();
}
// Incoming packet from network parsed and ready for decode.
@@ -902,10 +785,6 @@
return true;
}
-int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
- return receiver_.RemoveCodec(payload_type);
-}
-
int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
return receiver_.EnableNack(max_nack_list_size);
}
@@ -951,52 +830,4 @@
return new AudioCodingModuleImpl(config);
}
-int AudioCodingModule::NumberOfCodecs() {
- return static_cast<int>(acm2::RentACodec::NumberOfCodecs());
-}
-
-int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
- auto codec_id = acm2::RentACodec::CodecIdFromIndex(list_id);
- if (!codec_id)
- return -1;
- auto ci = acm2::RentACodec::CodecInstById(*codec_id);
- if (!ci)
- return -1;
- *codec = *ci;
- return 0;
-}
-
-int AudioCodingModule::Codec(const char* payload_name,
- CodecInst* codec,
- int sampling_freq_hz,
- size_t channels) {
- absl::optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
- payload_name, sampling_freq_hz, channels);
- if (ci) {
- *codec = *ci;
- return 0;
- } else {
- // We couldn't find a matching codec, so set the parameters to unacceptable
- // values and return.
- codec->plname[0] = '\0';
- codec->pltype = -1;
- codec->pacsize = 0;
- codec->rate = 0;
- codec->plfreq = 0;
- return -1;
- }
-}
-
-int AudioCodingModule::Codec(const char* payload_name,
- int sampling_freq_hz,
- size_t channels) {
- absl::optional<acm2::RentACodec::CodecId> ci =
- acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
- channels);
- if (!ci)
- return -1;
- absl::optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
- return i ? *i : -1;
-}
-
} // namespace webrtc
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 4e262f7..446ac64 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -19,7 +19,6 @@
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/acm2/acm_receive_test.h"
#include "modules/audio_coding/acm2/acm_send_test.h"
-#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
@@ -190,12 +189,11 @@
// Set up L16 codec.
virtual void SetUpL16Codec() {
audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1);
- ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1));
- codec_.pltype = kPayloadType;
+ pac_size_ = 160;
}
virtual void RegisterCodec() {
- EXPECT_EQ(true, acm_->RegisterReceiveCodec(kPayloadType, *audio_format_));
+ acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}});
acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
kPayloadType, *audio_format_, absl::nullopt));
}
@@ -226,7 +224,7 @@
virtual void VerifyEncoding() {
int last_length = packet_cb_.last_payload_len_bytes();
- EXPECT_TRUE(last_length == 2 * codec_.pacsize || last_length == 0)
+ EXPECT_TRUE(last_length == 2 * pac_size_ || last_length == 0)
<< "Last encoded packet was " << last_length << " bytes.";
}
@@ -241,10 +239,8 @@
WebRtcRTPHeader rtp_header_;
AudioFrame input_frame_;
- // These two have to be kept in sync for now. In the future, we'll be able to
- // eliminate the CodecInst and keep only the SdpAudioFormat.
absl::optional<SdpAudioFormat> audio_format_;
- CodecInst codec_;
+ int pac_size_ = -1;
Clock* clock_;
};
@@ -343,7 +339,7 @@
// Also checks that the frame type is kAudioFrameSpeech.
TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
const int k10MsBlocksPerPacket = 3;
- codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100;
+ pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100;
audio_format_->parameters["ptime"] = "30";
RegisterCodec();
const int kLoops = 10;
@@ -363,7 +359,7 @@
TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) {
RegisterCodec(); // This registers the default codec.
uint32_t expected_ts = input_frame_.timestamp_;
- int blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
+ int blocks_per_packet = pac_size_ / (kSampleRateHz / 100);
// Encode 5 packets of the first codec type.
const int kNumPackets1 = 5;
for (int j = 0; j < kNumPackets1; ++j) {
@@ -373,14 +369,14 @@
}
EXPECT_EQ(j + 1, packet_cb_.num_calls());
EXPECT_EQ(expected_ts, packet_cb_.last_timestamp());
- expected_ts += codec_.pacsize;
+ expected_ts += pac_size_;
}
// Change codec.
- ASSERT_EQ(0, AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1));
audio_format_ = SdpAudioFormat("ISAC", kSampleRateHz, 1);
+ pac_size_ = 480;
RegisterCodec();
- blocks_per_packet = codec_.pacsize / (kSampleRateHz / 100);
+ blocks_per_packet = pac_size_ / (kSampleRateHz / 100);
// Encode another 5 packets.
const int kNumPackets2 = 5;
for (int j = 0; j < kNumPackets2; ++j) {
@@ -390,7 +386,7 @@
}
EXPECT_EQ(kNumPackets1 + j + 1, packet_cb_.num_calls());
EXPECT_EQ(expected_ts, packet_cb_.last_timestamp());
- expected_ts += codec_.pacsize;
+ expected_ts += pac_size_;
}
}
#endif
@@ -404,9 +400,8 @@
: public AudioCodingModuleTestOldApi {
protected:
void RegisterCngCodec(int rtp_payload_type) {
- EXPECT_EQ(true,
- acm_->RegisterReceiveCodec(
- rtp_payload_type, SdpAudioFormat("cn", kSampleRateHz, 1)));
+ acm_->SetReceiveCodecs({{kPayloadType, *audio_format_},
+ {rtp_payload_type, {"cn", kSampleRateHz, 1}}});
acm_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
AudioEncoderCngConfig config;
config.speech_encoder = std::move(*enc);
@@ -461,7 +456,7 @@
TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi,
TransportCallbackTestForComfortNoiseRegisterCngLast) {
const int k10MsBlocksPerPacket = 3;
- codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100;
+ pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100;
audio_format_->parameters["ptime"] = "30";
RegisterCodec();
const int kCngPayloadType = 105;
@@ -650,12 +645,11 @@
void RegisterCodec() override {
static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
audio_format_ = SdpAudioFormat("isac", kSampleRateHz, 1);
- AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
- codec_.pltype = kPayloadType;
+ pac_size_ = 480;
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
// registered in AudioCodingModuleTestOldApi::SetUp();
- EXPECT_EQ(true, acm_->RegisterReceiveCodec(kPayloadType, *audio_format_));
+ acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}});
acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
kPayloadType, *audio_format_, absl::nullopt));
}
@@ -755,15 +749,11 @@
}
void RegisterCodec() override {
- static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
- AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1);
- codec_.pltype = kPayloadType;
-
// Register iSAC codec in ACM, effectively unregistering the PCM16B codec
// registered in AudioCodingModuleTestOldApi::SetUp();
// Only register the decoder for now. The encoder is registered later.
- ASSERT_EQ(true, acm_->RegisterReceiveCodec(codec_.pltype,
- CodecInstToSdp(codec_)));
+ static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz");
+ acm_->SetReceiveCodecs({{kPayloadType, {"ISAC", kSampleRateHz, 1}}});
}
void StartThreads() {
@@ -1085,7 +1075,17 @@
"bd44bf97e7899186532f91235cef444d",
"9d092dbc96e7ef6870b78c1056e87315"),
factory, [](AudioCodingModule* acm) {
- acm->RegisterReceiveCodec(0, {"MockPCMu", 8000, 1});
+ acm->SetReceiveCodecs({{0, {"MockPCMu", 8000, 1}},
+ {103, {"ISAC", 16000, 1}},
+ {104, {"ISAC", 32000, 1}},
+ {93, {"L16", 8000, 1}},
+ {94, {"L16", 16000, 1}},
+ {95, {"L16", 32000, 1}},
+ {8, {"PCMA", 8000, 1}},
+ {102, {"ILBC", 8000, 1}},
+ {13, {"CN", 8000, 1}},
+ {98, {"CN", 16000, 1}},
+ {99, {"CN", 32000, 1}}});
});
}
#endif
@@ -1746,11 +1746,11 @@
}
TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
- CodecInst codec_inst;
- codec_inst.channels = 1;
- codec_inst.pacsize = 160;
- codec_inst.pltype = 0;
- AudioEncoderPcmU encoder(codec_inst);
+ AudioEncoderPcmU::Config config;
+ config.frame_size_ms = 20;
+ config.num_channels = 1;
+ config.payload_type = 0;
+ AudioEncoderPcmU encoder(config);
auto mock_encoder = absl::make_unique<MockAudioEncoder>();
// Set expectations on the mock encoder and also delegate the calls to the
// real encoder.
@@ -1777,7 +1777,7 @@
uint32_t, rtc::ArrayView<const int16_t>, rtc::Buffer*)>(
&AudioEncoderPcmU::Encode)));
ASSERT_NO_FATAL_FAILURE(
- SetUpTestExternalEncoder(std::move(mock_encoder), codec_inst.pltype));
+ SetUpTestExternalEncoder(std::move(mock_encoder), config.payload_type));
Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
50, test::AcmReceiveTestOldApi::kMonoOutput);
}
diff --git a/modules/audio_coding/acm2/rent_a_codec.cc b/modules/audio_coding/acm2/rent_a_codec.cc
deleted file mode 100644
index bfddc42b..0000000
--- a/modules/audio_coding/acm2/rent_a_codec.cc
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2015 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 "modules/audio_coding/acm2/rent_a_codec.h"
-
-#include <memory>
-#include <utility>
-
-#include "rtc_base/logging.h"
-#include "modules/audio_coding/acm2/acm_codec_database.h"
-
-namespace webrtc {
-namespace acm2 {
-
-absl::optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
- const char* payload_name,
- int sampling_freq_hz,
- size_t channels) {
- return CodecIdFromIndex(
- ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels));
-}
-
-absl::optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
- absl::optional<int> mi = CodecIndexFromId(codec_id);
- return mi ? absl::optional<CodecInst>(ACMCodecDB::database_[*mi])
- : absl::nullopt;
-}
-
-absl::optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
- const CodecInst& codec_inst) {
- return CodecIdFromIndex(ACMCodecDB::CodecNumber(codec_inst));
-}
-
-absl::optional<CodecInst> RentACodec::CodecInstByParams(
- const char* payload_name,
- int sampling_freq_hz,
- size_t channels) {
- absl::optional<CodecId> codec_id =
- CodecIdByParams(payload_name, sampling_freq_hz, channels);
- if (!codec_id)
- return absl::nullopt;
- absl::optional<CodecInst> ci = CodecInstById(*codec_id);
- RTC_DCHECK(ci);
-
- // Keep the number of channels from the function call. For most codecs it
- // will be the same value as in default codec settings, but not for all.
- ci->channels = channels;
-
- return ci;
-}
-
-absl::optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
- CodecId codec_id,
- size_t num_channels) {
- absl::optional<int> i = CodecIndexFromId(codec_id);
- if (!i)
- return absl::nullopt;
- const NetEqDecoder ned = ACMCodecDB::neteq_decoders_[*i];
- return (ned == NetEqDecoder::kDecoderOpus && num_channels == 2)
- ? NetEqDecoder::kDecoderOpus_2ch
- : ned;
-}
-
-} // namespace acm2
-} // namespace webrtc
diff --git a/modules/audio_coding/acm2/rent_a_codec.h b/modules/audio_coding/acm2/rent_a_codec.h
deleted file mode 100644
index 2cf1c6e..0000000
--- a/modules/audio_coding/acm2/rent_a_codec.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2015 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 MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
-#define MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
-
-#include <stddef.h>
-#include <map>
-#include <memory>
-
-#include "absl/types/optional.h"
-#include "api/array_view.h"
-#include "api/audio_codecs/audio_decoder.h"
-#include "api/audio_codecs/audio_encoder.h"
-#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
-#include "rtc_base/constructormagic.h"
-#include "rtc_base/scoped_ref_ptr.h"
-
-namespace webrtc {
-
-struct CodecInst;
-class LockedIsacBandwidthInfo;
-
-namespace acm2 {
-
-struct RentACodec {
- enum class CodecId {
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
- kISAC,
-#endif
-#ifdef WEBRTC_CODEC_ISAC
- kISACSWB,
-#endif
- // Mono
- kPCM16B,
- kPCM16Bwb,
- kPCM16Bswb32kHz,
- // Stereo
- kPCM16B_2ch,
- kPCM16Bwb_2ch,
- kPCM16Bswb32kHz_2ch,
- // Mono
- kPCMU,
- kPCMA,
- // Stereo
- kPCMU_2ch,
- kPCMA_2ch,
-#ifdef WEBRTC_CODEC_ILBC
- kILBC,
-#endif
- kG722, // Mono
- kG722_2ch, // Stereo
-#ifdef WEBRTC_CODEC_OPUS
- kOpus, // Mono and stereo
-#endif
- kCNNB,
- kCNWB,
- kCNSWB,
-#ifdef ENABLE_48000_HZ
- kCNFB,
-#endif
- kAVT,
- kAVT16kHz,
- kAVT32kHz,
- kAVT48kHz,
-#ifdef WEBRTC_CODEC_RED
- kRED,
-#endif
- kNumCodecs, // Implementation detail. Don't use.
-
-// Set unsupported codecs to -1.
-#if !defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX)
- kISAC = -1,
-#endif
-#ifndef WEBRTC_CODEC_ISAC
- kISACSWB = -1,
-#endif
- // 48 kHz not supported, always set to -1.
- kPCM16Bswb48kHz = -1,
-#ifndef WEBRTC_CODEC_ILBC
- kILBC = -1,
-#endif
-#ifndef WEBRTC_CODEC_OPUS
- kOpus = -1, // Mono and stereo
-#endif
-#ifndef WEBRTC_CODEC_RED
- kRED = -1,
-#endif
-#ifndef ENABLE_48000_HZ
- kCNFB = -1,
-#endif
-
- kNone = -1
- };
-
- static inline size_t NumberOfCodecs() {
- return static_cast<size_t>(CodecId::kNumCodecs);
- }
-
- static inline absl::optional<int> CodecIndexFromId(CodecId codec_id) {
- const int i = static_cast<int>(codec_id);
- return i >= 0 && i < static_cast<int>(NumberOfCodecs())
- ? absl::optional<int>(i)
- : absl::nullopt;
- }
-
- static inline absl::optional<CodecId> CodecIdFromIndex(int codec_index) {
- return static_cast<size_t>(codec_index) < NumberOfCodecs()
- ? absl::optional<RentACodec::CodecId>(
- static_cast<RentACodec::CodecId>(codec_index))
- : absl::nullopt;
- }
-
- static absl::optional<CodecId> CodecIdByParams(const char* payload_name,
- int sampling_freq_hz,
- size_t channels);
- static absl::optional<CodecInst> CodecInstById(CodecId codec_id);
- static absl::optional<CodecId> CodecIdByInst(const CodecInst& codec_inst);
- static absl::optional<CodecInst> CodecInstByParams(const char* payload_name,
- int sampling_freq_hz,
- size_t channels);
-
- static inline bool IsPayloadTypeValid(int payload_type) {
- return payload_type >= 0 && payload_type <= 127;
- }
-
- static absl::optional<NetEqDecoder> NetEqDecoderFromCodecId(
- CodecId codec_id,
- size_t num_channels);
-};
-
-} // namespace acm2
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_