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_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