add flush encoder

build-demo.sh

@@ -10,7 +10,7 @@ echo "WORKPATH"=$WORKPATH
 
 rm -rf ${WORKPATH}/demo/mp4encoder.js ${WORKPATH}/demo/mp4encoder.wasm
 
-FFMPEG_ST=yes
+FFMPEG_ST=no
 
 EMSDK=/emsdk
 

demo_transcode_mp3/CMakeLists.txt

@@ -2,6 +2,14 @@ cmake_minimum_required(VERSION 3.10.0)
 project(transcode_mp3)
 
 set(CMAKE_CXX_STANDARD 17)
+set(CMAKE_CXX_FLAGS "\
+    -pthread \
+    -Wno-deprecated-declarations \
+    -Wno-pointer-sign \
+    -Wno-implicit-int-float-conversion \
+    -Wno-switch -Wno-parentheses\
+    -Qunused-arguments \
+")
 
 # set(CMAKE_EXECUTABLE_SUFFIX ".html") # 编译生成.html
 set( CMAKE_VERBOSE_MAKEFILE on )
@@ -24,17 +32,11 @@ add_subdirectory(audio)
 add_subdirectory(ffmbase)
 
 set_target_properties(transcode_mp3 PROPERTIES LINK_FLAGS "\
-    -Wno-deprecated-declarations \
-    -Wno-pointer-sign \
-    -Wno-implicit-int-float-conversion \
-    -Wno-switch -Wno-parentheses\
-    -Qunused-arguments \
     -s FORCE_FILESYSTEM=1 \
     -s WASM=1 \
     -s USE_SDL=2 \
     -s INVOKE_RUN=0 \
     -s EXIT_RUNTIME=1 \
-    -s MODULARIZE=1 \
     -s EXPORT_NAME=\"createTrancodeMp3\" \
     -s EXPORTED_FUNCTIONS=\"[_main,_malloc,_free]\" \
     -s EXPORTED_RUNTIME_METHODS=\"[FS, cwrap, ccall, setValue, writeAsciiToMemory]\" \
@@ -42,6 +44,10 @@ set_target_properties(transcode_mp3 PROPERTIES LINK_FLAGS "\
     -s ALLOW_MEMORY_GROWTH=1 \
     --pre-js ${PRE_PATH}/pre.js \
     --post-js ${POST_PATH}/post.js \
+    -pthread \
+    -s USE_PTHREADS=1 \
+    -s PROXY_TO_PTHREAD=1 \
+    -s PTHREAD_POOL_SIZE=8\
     ")
 
 target_link_libraries(${PROJECT_NAME} PRIVATE audio)

demo_transcode_mp3/audio/FFAudioDecoder.cpp

@@ -56,11 +56,8 @@ namespace FFM {
 
 	void FFAudioDecoder::startDecoder()
 	{
-		// std::thread t([this]() {
-		// 	decodeAudio();
-		// });
-		// t.join();
 		decodeAudio();
+
 	}
 
 	int FFAudioDecoder::decode(int *gotframe)
@@ -168,10 +165,14 @@ namespace FFM {
 					//}
 				}
 			}
-
 			av_packet_unref(m_pPkt);
 		}
 
+		if (m_audioEncoder) {
+			m_audioEncoder->flush_encoder();
+
+		}
+
 		printf("------------------count=%d\n", count);
 		/*if (m_audioEncoder) {
 			m_audioEncoder->closeEncoder();

demo_transcode_mp3/audio/FFAudioEncoder.cpp

@@ -834,12 +834,47 @@ namespace FFM {
 		printf("File duration:  %02d:%02d:%02d\n", thh, tmm, tss);
 	}
 
+	int FFAudioEncoder::writePacket(AVPacket *pkt)
+	{
+		int ret = -1;
+		av_packet_rescale_ts(pkt, m_pCodecCtx->time_base, m_pStream->time_base);
+
+		pkt->stream_index = m_pStream->index;
+		//AVStream *in_stream = m_pInStream;
+
+		////pkt.pts = av_rescale_q(m_swr->getLastTime(), m_swr->getTimeBase(), m_pFormatCtx->streams[pkt.stream_index]->time_base);
+
+		////
+		////pkt.pts = av_frame_get_best_effort_timestamp(filter_frame);
+		////pkt.dts = pkt.pts;
+
+		////pkt.pts = pkt.dts = m_frameIndex++ *( m_pCodecCtx->frame_size * 1000 / m_pCodecCtx->sample_rate);
+
+		pkt->pts = av_rescale_q_rnd(pkt->pts, m_pInCodeCtx->time_base, m_pStream->time_base, (AVRounding)(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
+		pkt->dts = av_rescale_q_rnd(pkt->dts, m_pInCodeCtx->time_base, m_pStream->time_base, (AVRounding)(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
+		pkt->duration = av_rescale_q(pkt->duration, m_pInCodeCtx->time_base, m_pStream->time_base);
+
+		//pkt.pos = -1;
+
+		printf("timestamp: %lf\n", pkt->pts*av_q2d(m_pStream->time_base));
+
+		printf("pkt.size: %d, frame.nb_samples: %d\npts: %ld, dts: %ld, duration: %ld\n", pkt->size,\
+			m_pStream->codec->frame_size,\
+			pkt->pts, pkt->dts, pkt->duration);
+
+		ret = av_interleaved_write_frame(m_pFormatCtx, pkt);
+		return ret;
+
+	}
+
 	int FFAudioEncoder::encodeAudio(AVCodecContext *inCtx, AVFrame *frame, uint64_t starttime)
 	{
 		uint64_t time = av_gettime() - starttime;
 		int ret = -1;
 
-		if (!m_pFormatCtx) return -1;
+		if (!m_pFormatCtx || !inCtx) return -1;
+
+		m_pInCodeCtx = inCtx;
 
 		AVFrame *outframe = nullptr;
 
@@ -886,39 +921,15 @@ namespace FFM {
 				}
 
 				AVPacket pkt;
+				av_init_packet(&pkt);
 				pkt.data = nullptr;
 				pkt.size = 0;
-				av_init_packet(&pkt);
+
 				while (avcodec_receive_packet(m_pCodecCtx, &pkt) >= 0)
 				{
 					printf("encoding.\n");
 
-					av_packet_rescale_ts(&pkt, m_pCodecCtx->time_base, m_pStream->time_base);
-
-					pkt.stream_index = m_pStream->index;
-					//AVStream *in_stream = m_pInStream;
-
-					////pkt.pts = av_rescale_q(m_swr->getLastTime(), m_swr->getTimeBase(), m_pFormatCtx->streams[pkt.stream_index]->time_base);
-
-					////
-					////pkt.pts = av_frame_get_best_effort_timestamp(filter_frame);
-					////pkt.dts = pkt.pts;
-
-					////pkt.pts = pkt.dts = m_frameIndex++ *( m_pCodecCtx->frame_size * 1000 / m_pCodecCtx->sample_rate);
-
-
-					pkt.pts = av_rescale_q_rnd(pkt.pts, inCtx->time_base, m_pStream->time_base, (AVRounding)(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
-					pkt.dts = av_rescale_q_rnd(pkt.dts, inCtx->time_base, m_pStream->time_base, (AVRounding)(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
-					pkt.duration = av_rescale_q(pkt.duration, inCtx->time_base, m_pStream->time_base);
-
-					pkt.pos = -1;
-
-					printf("timestamp: %lf\n", pkt.pts*av_q2d(m_pStream->time_base));
-
-					printf("pkt.size: %d, frame.nb_samples: %d\npts: %ld, dts: %ld, duration: %ld\n", pkt.size, m_pStream->codec->frame_size, pkt.pts, pkt.dts, pkt.duration);
-
-					m_pStream->duration = m_pInStream->duration;
-					av_interleaved_write_frame(m_pFormatCtx, &pkt);
+					writePacket(&pkt);
 					av_packet_unref(&pkt);
 				}
 
@@ -937,4 +948,38 @@ namespace FFM {
 		} while (r);
 		return ret;
 	}
-}
+
+	int FFAudioEncoder::flush_encoder()
+	{
+		int ret = -1;
+
+		for (;;) {
+			
+			AVPacket pkt;
+			av_init_packet(&pkt);
+
+			pkt.data = nullptr;
+			pkt.size = 0;
+			while ((ret = avcodec_receive_packet(m_pCodecCtx, &pkt)) == AVERROR(EAGAIN)) {
+				ret = avcodec_send_frame(m_pCodecCtx, NULL);
+				if (ret < 0) {
+					printf("encoding failed!\n");
+					exit(1);
+				}
+				printf(">>> have encode pkt\n");
+			}
+
+
+			if (ret == AVERROR_EOF || ret < 0) {
+				//av_packet_unref(&pkt);
+				break;
+			}
+
+			writePacket(&pkt);
+			av_packet_unref(&pkt);
+
+		}
+		printf(">>>> exit flush\n");
+		return ret;
+	}
+}
\ No newline at end of file

demo_transcode_mp3/audio/FFAudioEncoder.h

@@ -5,7 +5,7 @@
 
 /*
 .ogg	DE	y
-.opus	E	n
+.opus	E	y
 .aac	DE	y
 .mp3	DE	y
 .flac	DE	y
@@ -14,9 +14,9 @@
 .m4a	D	y
 .oga	E	y
 .mid		n
-.webm	DE	n
+.webm	DE	y
 .weba		n
-.amr	DE	n
+.amr	DE	y
 .au		DE	y
 .wma	D	y
 .aiff	DE	y
@@ -180,12 +180,17 @@ namespace FFM
 			m_filter = fileter;
 		}
 
+		int flush_encoder();
+
 	private:
 		int openFile(std::string &fileName);
 
 		int openCodec();
+
+		int writePacket(AVPacket *pkt);
 	private:
 		AVStream *m_pInStream = nullptr;
+		AVCodecContext *m_pInCodeCtx = nullptr;
 		AVFormatContext *m_pFormatCtx = nullptr;
 
 		AVOutputFormat* m_pOutFmt = nullptr;